NetAnim¶

NetAnim is a standalone, Qt4-based software executable that uses a trace file generated during an ns-3 simulation to display the topology and animate the packet flow between nodes.

Methodology¶

The class ns3::AnimationInterface is responsible for the creation the trace XML file. AnimationInterface uses the tracing infrastructure to track packet flows between nodes. AnimationInterface registers itself as a trace hook for tx and rx events before the simulation begins. When a packet is scheduled for transmission or reception, the corresponding tx and rx trace hooks in AnimationInterface are called. When the rx hooks are called, AnimationInterface will be aware of the two endpoints between which a packet has flowed, and adds this information to the trace file, in XML format along with the corresponding tx and rx timestamps. The XML format will be discussed in a later section. It is important to note that AnimationInterface records a packet only if the rx trace hooks are called. Every tx event must be matched by an rx event.

Downloading NetAnim¶

If NetAnim is not already available in the ns-3 package you downloaded, you can do the following:

Please ensure that you have installed mercurial. The latest version of NetAnim can be downloaded using mercurial with the following command:

$ hg clone http://code.nsnam.org/netanim

Building NetAnim¶

Prerequisites¶

Qt5 (5.4 and over) is required to build NetAnim. This can be obtained using the following ways:

For Ubuntu Linux distributions:

$ apt-get install qt5-default

For Red Hat/Fedora based distribution:

$ yum install qt5
$ yum install qt5-devel

For Mac/OSX, see http://qt.nokia.com/downloads/

Build steps¶

To build NetAnim use the following commands:

$ cd netanim
$ make clean
$ qmake NetAnim.pro
$ make

Note: qmake could be “qmake-qt5” in some systems

This should create an executable named “NetAnim” in the same directory:

 $ ls -l NetAnim
-rwxr-xr-x 1 john john 390395 2012-05-22 08:32 NetAnim

Usage¶

Using NetAnim is a two-step process

Step 1:Generate the animation XML trace file during simulation using “ns3::AnimationInterface” in the ns-3 code base.

Step 2:Load the XML trace file generated in Step 1 with the offline Qt4-based animator named NetAnim.

Step 1: Generate XML animation trace file¶

The class “AnimationInterface” under “src/netanim” uses underlying ns-3 trace sources to construct a timestamped ASCII file in XML format.

Examples are found under src/netanim/examples Example:

$ ./waf -d debug configure --enable-examples
$ ./waf --run "dumbbell-animation"

The above will create an XML file dumbbell-animation.xml

Mandatory¶

AnimationInterface anim ("animation.xml");  // where "animation.xml" is any arbitrary filename

[for versions before ns-3.13 you also have to use the line “anim.SetXMLOutput() to set the XML mode and also use anim.StartAnimation();]

Optional¶

The following are optional but useful steps:

// Step 1
anim.SetMobilityPollInterval (Seconds (1));

AnimationInterface records the position of all nodes every 250 ms by default. The statement above sets the periodic interval at which AnimationInterface records the position of all nodes. If the nodes are expected to move very little, it is useful to set a high mobility poll interval to avoid large XML files.

// Step 2
anim.SetConstantPosition (Ptr< Node > n, double x, double y);

AnimationInterface requires that the position of all nodes be set. In ns-3 this is done by setting an associated MobilityModel. “SetConstantPosition” is a quick way to set the x-y coordinates of a node which is stationary.

// Step 3
anim.SetStartTime (Seconds(150)); and anim.SetStopTime (Seconds(150));

AnimationInterface can generate large XML files. The above statements restricts the window between which AnimationInterface does tracing. Restricting the window serves to focus only on relevant portions of the simulation and creating manageably small XML files

// Step 4
AnimationInterface anim ("animation.xml", 50000);

Using the above constructor ensures that each animation XML trace file has only 50000 packets. For example, if AnimationInterface captures 150000 packets, using the above constructor splits the capture into 3 files

// Step 5
anim.EnablePacketMetadata (true);

With the above statement, AnimationInterface records the meta-data of each packet in the xml trace file. Metadata can be used by NetAnim to provide better statistics and filter, along with providing some brief information about the packet such as TCP sequence number or source & destination IP address during packet animation.

CAUTION: Enabling this feature will result in larger XML trace files. Please do NOT enable this feature when using Wimax links.

// Step 6
anim.UpdateNodeDescription (5, "Access-point");

With the above statement, AnimationInterface assigns the text “Access-point” to node 5.

// Step 7
anim.UpdateNodeSize (6, 1.5, 1.5);

With the above statement, AnimationInterface sets the node size to scale by 1.5. NetAnim automatically scales the graphics view to fit the oboundaries of the topology. This means that NetAnim, can abnormally scale a node’s size too high or too low. Using AnimationInterface::UpdateNodeSize allows you to overwrite the default scaling in NetAnim and use your own custom scale.

// Step 8
anim.UpdateNodeCounter (89, 7, 3.4);

With the above statement, AnimationInterface sets the counter with Id == 89, associated with Node 7 with the value 3.4. The counter with Id 89 is obtained using AnimationInterface::AddNodeCounter. An example usage for this is in src/netanim/examples/resource-counters.cc.

Step 2: Loading the XML in NetAnim¶

Here is a video illustrating this http://www.youtube.com/watch?v=tz_hUuNwFDs

Wiki¶

For detailed instructions on installing “NetAnim”, F.A.Qs and loading the XML trace file (mentioned earlier) using NetAnim please refer: http://www.nsnam.org/wiki/NetAnim

Design documentation¶

Overview¶

The Antenna module provides:

Angles¶

The Angles class holds information about an angle in 3D space using spherical coordinates in radian units. Specifically, it uses the azimuth-inclination convention, where

Multiple constructors are present, supporting the most common ways to encode information on a direction. A static boolean variable allows the user to decide whether angles should be printed in radian or degree units.

A number of angle-related utilities are offered, such as radians/degree conversions, for both scalars and vectors, and angle wrapping.

AntennaModel¶

The AntennaModel uses the coordinate system adopted in [Balanis] and depicted in Figure Coordinate system of the AntennaModel. This system is obtained by translating the Cartesian coordinate system used by the ns-3 MobilityModel into the new origin o which is the location of the antenna, and then transforming the coordinates of every generic point p of the space from Cartesian coordinates (x,y,z) into spherical coordinates (r, \theta,\phi). The antenna model neglects the radial component r, and only considers the angle components (\theta, \phi). An antenna radiation pattern is then expressed as a mathematical function g(\theta, \phi) \longrightarrow \mathcal{R} that returns the gain (in dB) for each possible direction of transmission/reception. All angles are expressed in radians.

Single antenna models¶

In this section we describe the antenna radiation pattern models that are included within the antenna module.

IsotropicAntennaModel¶

This antenna radiation pattern model provides a unitary gain (0 dB) for all direction.

CosineAntennaModel¶

This is the cosine model described in [Chunjian]: the antenna gain is determined as:

g(\phi, \theta) = \cos^{n} \left(\frac{\phi - \phi_{0}}{2} \right)

where \phi_{0} is the azimuthal orientation of the antenna (i.e., its direction of maximum gain) and the exponential

n = -\frac{3}{20 \log_{10} \left( \cos \frac{\phi_{3dB}}{4} \right)}

determines the desired 3dB beamwidth \phi_{3dB}. Note that this radiation pattern is independent of the inclination angle \theta.

A major difference between the model of [Chunjian] and the one implemented in the class CosineAntennaModel is that only the element factor (i.e., what described by the above formulas) is considered. In fact, [Chunjian] also considered an additional antenna array factor. The reason why the latter is excluded is that we expect that the average user would desire to specify a given beamwidth exactly, without adding an array factor at a latter stage which would in practice alter the effective beamwidth of the resulting radiation pattern.

ParabolicAntennaModel¶

This model is based on the parabolic approximation of the main lobe radiation pattern. It is often used in the context of cellular system to model the radiation pattern of a cell sector, see for instance [R4-092042a] and [Calcev]. The antenna gain in dB is determined as:

g_{dB}(\phi, \theta) = -\min \left( 12 \left(\frac{\phi - \phi_{0}}{\phi_{3dB}} \right)^2, A_{max} \right)

where \phi_{0} is the azimuthal orientation of the antenna (i.e., its direction of maximum gain), \phi_{3dB} is its 3 dB beamwidth, and A_{max} is the maximum attenuation in dB of the antenna. Note that this radiation pattern is independent of the inclination angle \theta.

ThreeGppAntennaModel¶

This model implements the antenna element described in [38901]. Parameters are fixed from the technical report, thus no attributes nor setters are provided. The model is largely based on the ParabolicAntennaModel.

Phased Array Model¶

The class PhasedArrayModel has been created with flexibility in mind. It abstracts the basic idea of a Phased Antenna Array (PAA) by removing any constraint on the position of each element, and instead generalizes the concept of steering and beamforming vectors, solely based on the generalized location of the antenna elements. For details on Phased Array Antennas see for instance [Mailloux].

Derived classes must implement the following functions:

The class PhasedArrayModel also assumes that all antenna elements are equal, a typical key assumption which allows to model the PAA field pattern as the sum of the array factor, given by the geometry of the location of the antenna elements, and the element field pattern. Any class derived from AntennaModel is a valid antenna element for the PhasedArrayModel, allowing for a great flexibility of the framework.

UniformPlanarArray¶

The class UniformPlanarArray is a generic implementation of Uniform Planar Arrays (UPAs), supporting rectangular and linear regular lattices. It loosely follows the implementation described in the 3GPP TR 38.901 [38901], considering only a single a single panel, i.e., N_{g} = M_{g} = 1.

By default, the array is orthogonal to the x-axis, pointing towards the positive direction, but the orientation can be changed through the attributes “BearingAngle”, which adjusts the azimuth angle, and “DowntiltAngle”, which adjusts the elevation angle. The slant angle is instead fixed and assumed to be 0.

The number of antenna elements in the vertical and horizontal directions can be configured through the attributes “NumRows” and “NumColumns”, while the spacing between the horizontal and vertical elements can be configured through the attributes “AntennaHorizontalSpacing” and “AntennaVerticalSpacing”.

The polarization of each antenna element in the array is determined by the polarization slant angle through the attribute “PolSlantAngle”, as described in [38901] (i.e., {\zeta}).

[Balanis]

C.A. Balanis, “Antenna Theory - Analysis and Design”, Wiley, 2nd Ed.

[Chunjian]

Li Chunjian, “Efficient Antenna Patterns for Three-Sector WCDMA Systems”, Master of Science Thesis, Chalmers University of Technology, Göteborg, Sweden, 2003

[Calcev]

George Calcev and Matt Dillon, “Antenna Tilt Control in CDMA Networks”, in Proc. of the 2nd Annual International Wireless Internet Conference (WICON), 2006

[R4-092042a]

3GPP TSG RAN WG4 (Radio) Meeting #51, R4-092042, Simulation assumptions and parameters for FDD HeNB RF requirements.

[38901]

3GPP. 2018. TR 38.901, Study on channel model for frequencies from 0.5 to 100 GHz, V15.0.0. (2018-06).

[Mailloux]

Robert J. Mailloux, “Phased Array Antenna Handbook”, Artech House, 2nd Ed.

User Documentation¶

The antenna modeled can be used with all the wireless technologies and physical layer models that support it. Currently, this includes the physical layer models based on the SpectrumPhy. Please refer to the documentation of each of these models for details.

Testing Documentation¶

In this section we describe the test suites included with the antenna module that verify its correct functionality.

Angles¶

The unit test suite angles verifies that the Angles class is constructed properly by correct conversion from 3D Cartesian coordinates according to the available methods (construction from a single vector and from a pair of vectors). For each method, several test cases are provided that compare the values (\phi, \theta) determined by the constructor to known reference values. The test passes if for each case the values are equal to the reference up to a tolerance of 10^{-10} which accounts for numerical errors.

DegreesToRadians¶

The unit test suite degrees-radians verifies that the methods DegreesToRadians and RadiansToDegrees work properly by comparing with known reference values in a number of test cases. Each test case passes if the comparison is equal up to a tolerance of 10^{-10} which accounts for numerical errors.

IsotropicAntennaModel¶

The unit test suite isotropic-antenna-model checks that the IsotropicAntennaModel class works properly, i.e., returns always a 0dB gain regardless of the direction.

CosineAntennaModel¶

The unit test suite cosine-antenna-model checks that the CosineAntennaModel class works properly. Several test cases are provided that check for the antenna gain value calculated at different directions and for different values of the orientation, the reference gain and the beamwidth. The reference gain is calculated by hand. Each test case passes if the reference gain in dB is equal to the value returned by CosineAntennaModel within a tolerance of 0.001, which accounts for the approximation done for the calculation of the reference values.

ParabolicAntennaModel¶

The unit test suite parabolic-antenna-model checks that the ParabolicAntennaModel class works properly. Several test cases are provided that check for the antenna gain value calculated at different directions and for different values of the orientation, the maximum attenuation and the beamwidth. The reference gain is calculated by hand. Each test case passes if the reference gain in dB is equal to the value returned by ParabolicAntennaModel within a tolerance of 0.001, which accounts for the approximation done for the calculation of the reference values.

Model Description¶

The source code for the AODV model lives in the directory src/aodv.

Design¶

Class ns3::aodv::RoutingProtocol implements all functionality of service packet exchange and inherits from ns3::Ipv4RoutingProtocol. The base class defines two virtual functions for packet routing and forwarding. The first one, ns3::aodv::RouteOutput, is used for locally originated packets, and the second one, ns3::aodv::RouteInput, is used for forwarding and/or delivering received packets.

Protocol operation depends on many adjustable parameters. Parameters for this functionality are attributes of ns3::aodv::RoutingProtocol. Parameter default values are drawn from the RFC and allow the enabling/disabling protocol features, such as broadcasting HELLO messages, broadcasting data packets and so on.

AODV discovers routes on demand. Therefore, the AODV model buffers all packets while a route request packet (RREQ) is disseminated. A packet queue is implemented in aodv-rqueue.cc. A smart pointer to the packet, ns3::Ipv4RoutingProtocol::ErrorCallback, ns3::Ipv4RoutingProtocol::UnicastForwardCallback, and the IP header are stored in this queue. The packet queue implements garbage collection of old packets and a queue size limit.

The routing table implementation supports garbage collection of old entries and state machine, defined in the standard. It is implemented as a STL map container. The key is a destination IP address.

Some elements of protocol operation aren’t described in the RFC. These elements generally concern cooperation of different OSI model layers. The model uses the following heuristics:

The layer 2 feedback implementation relies on the TxErrHeader trace source, currently supported in AdhocWifiMac only.

Scope and Limitations¶

The model is for IPv4 only. The following optional protocol optimizations are not implemented:

These techniques require direct access to IP header, which contradicts the assertion from the AODV RFC that AODV works over UDP. This model uses UDP for simplicity, hindering the ability to implement certain protocol optimizations. The model doesn’t use low layer raw sockets because they are not portable.

Future Work¶

No announced plans.

Model Description¶

The model is a part of the applications library. The HTTP model is based on a commonly used 3GPP model in standardization [4].

Design¶

This traffic generator simulates web browsing traffic using the Hypertext Transfer Protocol (HTTP). It consists of one or more ThreeGppHttpClient applications which connect to a ThreeGppHttpServer application. The client models a web browser which requests web pages to the server. The server is then responsible to serve the web pages as requested. Please refer to ThreeGppHttpClientHelper and ThreeGppHttpServerHelper for usage instructions.

Technically speaking, the client transmits request objects to demand a service from the server. Depending on the type of request received, the server transmits either:

The main and embedded object sizes are illustrated in figures 3GPP HTTP main object size histogram and 3GPP HTTP embedded object size histogram.

3GPP HTTP server description¶

3GPP HTTP server is a model application which simulates the traffic of a web server. This application works in conjunction with ThreeGppHttpClient applications.

The application works by responding to requests. Each request is a small packet of data which contains ThreeGppHttpHeader. The value of the content type field of the header determines the type of object that the client is requesting. The possible type is either a main object or an embedded object.

The application is responsible to generate the right type of object and send it back to the client. The size of each object to be sent is randomly determined (see ThreeGppHttpVariables). Each object may be sent as multiple packets due to limited socket buffer space.

To assist with the transmission, the application maintains several instances of ThreeGppHttpServerTxBuffer. Each instance keeps track of the object type to be served and the number of bytes left to be sent.

The application accepts connection request from clients. Every connection is kept open until the client disconnects.

Maximum transmission unit (MTU) size is configurable in ThreeGppHttpServer or in ThreeGppHttpVariables. By default, the low variant is 536 bytes and high variant is 1460 bytes. The default values are set with the intention of having a TCP header (size of which is 40 bytes) added in the packet in such way that lower layers can avoid splitting packets. The change of MTU sizes affects all TCP sockets after the server application has started. It is mainly visible in sizes of packets received by ThreeGppHttpClient applications.

3GPP HTTP client description¶

3GPP HTTP client is a model application which simulates the traffic of a web browser. This application works in conjunction with an ThreeGppHttpServer application.

In summary, the application works as follows.

References¶

Many aspects of the traffic are randomly determined by ThreeGppHttpVariables. A separate instance of this object is used by the HTTP server and client applications. These characteristics are based on a legacy 3GPP specification. The description can be found in the following references:

[1] 3GPP TR 25.892, “Feasibility Study for Orthogonal Frequency Division Multiplexing (OFDM) for UTRAN enhancement”

[2] IEEE 802.16m, “Evaluation Methodology Document (EMD)”, IEEE 802.16m-08/004r5, July 2008.

[3] NGMN Alliance, “NGMN Radio Access Performance Evaluation Methodology”, v1.0, January 2008.

[4] 3GPP2-TSGC5, “HTTP, FTP and TCP models for 1xEV-DV simulations”, 2001.

Usage¶

The three-gpp-http-example can be referenced to see basic usage of the HTTP applications. In summary, using the ThreeGppHttpServerHelper and ThreeGppHttpClientHelper allow the user to easily install ThreeGppHttpServer and ThreeGppHttpClient applications to nodes. The helper objects can be used to configure attribute values for the client and server objects, but not for the ThreeGppHttpVariables object. Configuration of variables is done by modifying attributes of ThreeGppHttpVariables, which should be done prior to helpers installing applications to nodes.

The client and server provide a number of ns-3 trace sources such as “Tx”, “Rx”, “RxDelay”, and “StateTransition” on the server side, and a large number on the client side (“ConnectionEstablished”, “ConnectionClosed”,”TxMainObjectRequest”, “TxEmbeddedObjectRequest”, “RxMainObjectPacket”, “RxMainObject”, “RxEmbeddedObjectPacket”, “RxEmbeddedObject”, “Rx”, “RxDelay”, “RxRtt”, “StateTransition”).

Building the 3GPP HTTP applications¶

Building the applications does not require any special steps to be taken. It suffices to enable the applications module.

Examples¶

For an example demonstrating HTTP applications run:

$ ./waf --run 'three-gpp-http-example'

By default, the example will print out the web page requests of the client and responses of the server and client receiving content packets by using LOG_INFO of ThreeGppHttpServer and ThreeGppHttpClient.

Tests¶

For testing HTTP applications, three-gpp-http-client-server-test is provided. Run:

$ ./test.py -s three-gpp-http-client-server-test

The test consists of simple Internet nodes having HTTP server and client applications installed. Multiple variant scenarios are tested: delay is 3ms, 30ms or 300ms, bit error rate 0 or 5.0*10^(-6), MTU size 536 or 1460 bytes and either IPV4 or IPV6 is used. A simulation with each combination of these parameters is run multiple times to verify functionality with different random variables.

Test cases themselves are rather simple: test verifies that HTTP object packet bytes sent match total bytes received by the client, and that ThreeGppHttpHeader matches the expected packet.

Model Description¶

The model relies on building an external BRITE library, and then building some ns-3 helpers that call out to the library. The source code for the ns-3 helpers lives in the directory src/brite/helper.

Design¶

To generate the BRITE topology, ns-3 helpers call out to the external BRITE library, and using a standard BRITE configuration file, the BRITE code builds a graph with nodes and edges according to this configuration file. Please see the BRITE documentation or the example configuration files in src/brite/examples/conf_files to get a better grasp of BRITE configuration options. The graph built by BRITE is returned to ns-3, and a ns-3 implementation of the graph is built. Leaf nodes for each AS are available for the user to either attach custom topologies or install ns-3 applications directly.

References¶

[1]

Alberto Medina, Anukool Lakhina, Ibrahim Matta, and John Byers. BRITE: An Approach to Universal Topology Generation. In Proceedings of the International Workshop on Modeling, Analysis and Simulation of Computer and Telecommunications Systems- MASCOTS ‘01, Cincinnati, Ohio, August 2001.

Usage¶

The brite-generic-example can be referenced to see basic usage of the BRITE interface. In summary, the BriteTopologyHelper is used as the interface point by passing in a BRITE configuration file. Along with the configuration file a BRITE formatted random seed file can also be passed in. If a seed file is not passed in, the helper will create a seed file using ns-3’s UniformRandomVariable. Once the topology has been generated by BRITE, BuildBriteTopology() is called to create the ns-3 representation. Next IP Address can be assigned to the topology using either AssignIpv4Addresses() or AssignIpv6Addresses(). It should be noted that each point-to-point link in the topology will be treated as a new network therefore for IPV4 a /30 subnet should be used to avoid wasting a large amount of the available address space.

Example BRITE configuration files can be found in /src/brite/examples/conf_files/. ASBarbasi and ASWaxman are examples of AS only topologies. The RTBarabasi and RTWaxman files are examples of router only topologies. Finally the TD_ASBarabasi_RTWaxman configuration file is an example of a Hierarchical topology that uses the Barabasi-Albert model for the AS level and the Waxman model for each of the router level topologies. Information on the BRITE parameters used in these files can be found in the BRITE user manual.

Building BRITE Integration¶

The first step is to download and build the ns-3 specific BRITE repository:

$ hg clone http://code.nsnam.org/BRITE
$ cd BRITE
$ make

This will build BRITE and create a library, libbrite.so, within the BRITE directory.

Once BRITE has been built successfully, we proceed to configure ns-3 with BRITE support. Change to your ns-3 directory:

$ ./waf configure --with-brite=/your/path/to/brite/source --enable-examples

Make sure it says ‘enabled’ beside ‘BRITE Integration’. If it does not, then something has gone wrong. Either you have forgotten to build BRITE first following the steps above, or ns-3 could not find your BRITE directory.

Next, build ns-3:

$ ./waf

Examples¶

For an example demonstrating BRITE integration run:

$ ./waf --run 'brite-generic-example'

By enabling the verbose parameter, the example will print out the node and edge information in a similar format to standard BRITE output. There are many other command-line parameters including confFile, tracing, and nix, described below:

The generic BRITE example also support visualization using pyviz, assuming python bindings in ns-3 are enabled:

$ ./waf --run brite-generic-example --vis

Simulations involving BRITE can also be used with MPI. The total number of MPI instances is passed to the BRITE topology helper where a modulo divide is used to assign the nodes for each AS to a MPI instance. An example can be found in src/brite/examples:

$ mpirun -np 2 ./waf --run brite-MPI-example

Please see the ns-3 MPI documentation for information on setting up MPI with ns-3.

Design documentation¶

Overview¶

The Buildings module provides:

The models have been designed with LTE in mind, though their implementation is in fact independent from any LTE-specific code, and can be used with other ns-3 wireless technologies as well (e.g., wifi, wimax).

The HybridBuildingsPropagationLossModel pathloss model included is obtained through a combination of several well known pathloss models in order to mimic different environmental scenarios such as urban, suburban and open areas. Moreover, the model considers both outdoor and indoor indoor and outdoor communication has to be included since HeNB might be installed either within building and either outside. In case of indoor communication, the model has to consider also the type of building in outdoor <-> indoor communication according to some general criteria such as the wall penetration losses of the common materials; moreover it includes some general configuration for the internal walls in indoor communications.

The OhBuildingsPropagationLossModel pathloss model has been created for simplifying the previous one removing the thresholds for switching from one model to other. For doing this it has been used only one propagation model from the one available (i.e., the Okumura Hata). The presence of building is still considered in the model; therefore all the considerations of above regarding the building type are still valid. The same consideration can be done for what concern the environmental scenario and frequency since both of them are parameters of the model considered.

The Building class¶

The model includes a specific class called Building which contains a ns3 Box class for defining the dimension of the building. In order to implements the characteristics of the pathloss models included, the Building class supports the following attributes:

The Building class is based on the following assumptions:

The MobilityBuildingInfo class¶

The MobilityBuildingInfo class, which inherits from the ns3 class Object, is in charge of maintaining information about the position of a node with respect to building. The information managed by MobilityBuildingInfo is:

The class MobilityBuildingInfo is used by BuildingsPropagationLossModel class, which inherits from the ns3 class PropagationLossModel and manages the pathloss computation of the single components and their composition according to the nodes’ positions. Moreover, it implements also the shadowing, that is the loss due to obstacles in the main path (i.e., vegetation, buildings, etc.).

It is to be noted that, MobilityBuildingInfo can be used by any other propagation model. However, based on the information at the time of this writing, only the ones defined in the building module are designed for considering the constraints introduced by the buildings.

ItuR1238PropagationLossModel¶

This class implements a building-dependent indoor propagation loss model based on the ITU P.1238 model, which includes losses due to type of building (i.e., residential, office and commercial). The analytical expression is given in the following.

L_\mathrm{total} = 20\log f + N\log d + L_f(n)- 28 [dB]

where:

BuildingsPropagationLossModel¶

The BuildingsPropagationLossModel provides an additional set of building-dependent pathloss model elements that are used to implement different pathloss logics. These pathloss model elements are described in the following subsections.

External Wall Loss (EWL)¶

This component models the penetration loss through walls for indoor to outdoor communications and vice-versa. The values are taken from the [cost231] model.

Internal Walls Loss (IWL)¶

This component models the penetration loss occurring in indoor-to-indoor communications within the same building. The total loss is calculated assuming that each single internal wall has a constant penetration loss L_{siw}, and approximating the number of walls that are penetrated with the manhattan distance (in number of rooms) between the transmitter and the receiver. In detail, let x_1, y_1, x_2, y_2 denote the room number along the x and y axis respectively for user 1 and 2; the total loss L_{IWL} is calculated as

L_{IWL} = L_{siw} (|x_1 -x_2| + |y_1 - y_2|)

Height Gain Model (HG)¶

This component model the gain due to the fact that the transmitting device is on a floor above the ground. In the literature [turkmani] this gain has been evaluated as about 2 dB per floor. This gain can be applied to all the indoor to outdoor communications and vice-versa.

Shadowing Model¶

The shadowing is modeled according to a log-normal distribution with variable standard deviation as function of the relative position (indoor or outdoor) of the MobilityModel instances involved. One random value is drawn for each pair of MobilityModels, and stays constant for that pair during the whole simulation. Thus, the model is appropriate for static nodes only.

The model considers that the mean of the shadowing loss in dB is always 0. For the variance, the model considers three possible values of standard deviation, in detail:

The simulator generates a shadowing value per each active link according to nodes’ position the first time the link is used for transmitting. In case of transmissions from outdoor nodes to indoor ones, and vice-versa, the standard deviation (\sigma_\mathrm{IO}) has to be calculated as the square root of the sum of the quadratic values of the standard deviatio in case of outdoor nodes and the one for the external walls penetration. This is due to the fact that that the components producing the shadowing are independent of each other; therefore, the variance of a distribution resulting from the sum of two independent normal ones is the sum of the variances.

X \sim N(\mu,\sigma^2) \mbox{ and } Y \sim N(\nu,\tau^2)

Z = X + Y \sim Z (\mu + \nu, \sigma^2 + \tau^2)

\Rightarrow \sigma_\mathrm{IO} = \sqrt{\sigma_\mathrm{O}^2 + \sigma_\mathrm{W}^2}

Pathloss logics¶

In the following we describe the different pathloss logic that are implemented by inheriting from BuildingsPropagationLossModel.

HybridBuildingsPropagationLossModel¶

The HybridBuildingsPropagationLossModel pathloss model included is obtained through a combination of several well known pathloss models in order to mimic different outdoor and indoor scenarios, as well as indoor-to-outdoor and outdoor-to-indoor scenarios. In detail, the class HybridBuildingsPropagationLossModel integrates the following pathloss models:

The following pseudo-code illustrates how the different pathloss model elements described above are integrated in HybridBuildingsPropagationLossModel:

if (txNode is outdoor)


  then


    if (rxNode is outdoor)


      then


        if (distance > 1 km)


          then


            if (rxNode or txNode is below the rooftop)


              then


                L = I1411


              else


                L = OH


          else


            L = I1411


      else (rxNode is indoor)


        if (distance > 1 km)


          then


            if (rxNode or txNode is below the rooftop)


              L = I1411 + EWL + HG


            else


              L = OH + EWL + HG


          else


            L = I1411 + EWL + HG
else (txNode is indoor)


  if (rxNode is indoor)


    then


     if (same building)


        then


          L = I1238 + IWL


        else


          L = I1411 + 2*EWL


   else (rxNode is outdoor)


    if (distance > 1 km)


      then


        if (rxNode or txNode is below the rooftop)


              then


                L = I1411 + EWL + HG


              else


                L = OH + EWL + HG


      else


        L = I1411 + EWL

We note that, for the case of communication between two nodes below rooftop level with distance is greater then 1 km, we still consider the I1411 model, since OH is specifically designed for macro cells and therefore for antennas above the roof-top level.

For the ITU-R P.1411 model we consider both the LOS and NLoS versions. In particular, we considers the LoS propagation for distances that are shorted than a tunable threshold (m_itu1411NlosThreshold). In case on NLoS propagation, the over the roof-top model is taken in consideration for modeling both macro BS and SC. In case on NLoS several parameters scenario dependent have been included, such as average street width, orientation, etc. The values of such parameters have to be properly set according to the scenario implemented, the model does not calculate natively their values. In case any values is provided, the standard ones are used, apart for the height of the mobile and BS, which instead their integrity is tested directly in the code (i.e., they have to be greater then zero). In the following we give the expressions of the components of the model.

We also note that the use of different propagation models (OH, I1411, I1238 with their variants) in HybridBuildingsPropagationLossModel can result in discontinuities of the pathloss with respect to distance. A proper tuning of the attributes (especially the distance threshold attributes) can avoid these discontinuities. However, since the behavior of each model depends on several other parameters (frequency, node height, etc), there is no default value of these thresholds that can avoid the discontinuities in all possible configurations. Hence, an appropriate tuning of these parameters is left to the user.

OhBuildingsPropagationLossModel¶

The OhBuildingsPropagationLossModel class has been created as a simple means to solve the discontinuity problems of HybridBuildingsPropagationLossModel without doing scenario-specific parameter tuning. The solution is to use only one propagation loss model (i.e., Okumura Hata), while retaining the structure of the pathloss logic for the calculation of other path loss components (such as wall penetration losses). The result is a model that is free of discontinuities (except those due to walls), but that is less realistic overall for a generic scenario with buildings and outdoor/indoor users, e.g., because Okumura Hata is not suitable neither for indoor communications nor for outdoor communications below rooftop level.

In detail, the class OhBuildingsPropagationLossModel integrates the following pathloss models:

The following pseudo-code illustrates how the different pathloss model elements described above are integrated in OhBuildingsPropagationLossModel:

if (txNode is outdoor)


  then


    if (rxNode is outdoor)


      then


        L = OH


      else (rxNode is indoor)


        L = OH + EWL
else (txNode is indoor)


  if (rxNode is indoor)


    then


     if (same building)


        then


          L = OH + IWL


        else


          L = OH + 2*EWL


   else (rxNode is outdoor)


      L = OH + EWL

We note that OhBuildingsPropagationLossModel is a significant simplification with respect to HybridBuildingsPropagationLossModel, due to the fact that OH is used always. While this gives a less accurate model in some scenarios (especially below rooftop and indoor), it effectively avoids the issue of pathloss discontinuities that affects HybridBuildingsPropagationLossModel.

User Documentation¶

How to use buildings in a simulation¶

In this section we explain the basic usage of the buildings model within a simulation program.

Include the headers¶

Add this at the beginning of your simulation program:

#include <ns3/buildings-module.h>

Create a building¶

As an example, let’s create a residential 10 x 20 x 10 building:

double x_min = 0.0;
double x_max = 10.0;
double y_min = 0.0;
double y_max = 20.0;
double z_min = 0.0;
double z_max = 10.0;
Ptr<Building> b = CreateObject <Building> ();
b->SetBoundaries (Box (x_min, x_max, y_min, y_max, z_min, z_max));
b->SetBuildingType (Building::Residential);
b->SetExtWallsType (Building::ConcreteWithWindows);
b->SetNFloors (3);
b->SetNRoomsX (3);
b->SetNRoomsY (2);

This building has three floors and an internal 3 x 2 grid of rooms of equal size.

The helper class GridBuildingAllocator is also available to easily create a set of buildings with identical characteristics placed on a rectangular grid. Here’s an example of how to use it:

Ptr<GridBuildingAllocator>  gridBuildingAllocator;
gridBuildingAllocator = CreateObject<GridBuildingAllocator> ();
gridBuildingAllocator->SetAttribute ("GridWidth", UintegerValue (3));
gridBuildingAllocator->SetAttribute ("LengthX", DoubleValue (7));
gridBuildingAllocator->SetAttribute ("LengthY", DoubleValue (13));
gridBuildingAllocator->SetAttribute ("DeltaX", DoubleValue (3));
gridBuildingAllocator->SetAttribute ("DeltaY", DoubleValue (3));
gridBuildingAllocator->SetAttribute ("Height", DoubleValue (6));
gridBuildingAllocator->SetBuildingAttribute ("NRoomsX", UintegerValue (2));
gridBuildingAllocator->SetBuildingAttribute ("NRoomsY", UintegerValue (4));
gridBuildingAllocator->SetBuildingAttribute ("NFloors", UintegerValue (2));
gridBuildingAllocator->SetAttribute ("MinX", DoubleValue (0));
gridBuildingAllocator->SetAttribute ("MinY", DoubleValue (0));
gridBuildingAllocator->Create (6);

This will create a 3x2 grid of 6 buildings, each 7 x 13 x 6 m with 2 x 4 rooms inside and 2 foors; the buildings are spaced by 3 m on both the x and the y axis.

Setup nodes and mobility models¶

Nodes and mobility models are configured as usual, however in order to use them with the buildings model you need an additional call to BuildingsHelper::Install(), so as to let the mobility model include the information on their position w.r.t. the buildings. Here is an example:

MobilityHelper mobility;
mobility.SetMobilityModel ("ns3::ConstantPositionMobilityModel");
ueNodes.Create (2);
mobility.Install (ueNodes);
BuildingsHelper::Install (ueNodes);

It is to be noted that any mobility model can be used. However, the user is advised to make sure that the behavior of the mobility model being used is consistent with the presence of Buildings. For example, using a simple random mobility over the whole simulation area in presence of buildings might easily results in node moving in and out of buildings, regardless of the presence of walls.

One dedicated buildings-aware mobility model is the RandomWalk2dOutdoorMobilityModel. This class is similar to the RandomWalk2dMobilityModel but avoids placing the trajectory on a path that would intersect a building wall. If a boundary is encountered (either the bounding box or a building wall), the model rebounds with a random direction and speed that ensures that the trajectory stays outside the buildings. An example program that demonstrates the use of this model is the src/buildings/examples/outdoor-random-walk-example.cc which has an associated shell script to plot the traces generated. Another example program demonstrates how this outdoor mobility model can be used as the basis of a group mobility model, with the outdoor buildings-aware model serving as the parent or reference mobility model, and with additional nodes defining a child mobility model providing the offset from the reference mobility model. This example, src/buildings/example/outdoor-group-mobility-example.cc, also has an associated shell script (outdoor-group-mobility-animate.sh) that can be used to generate an animated GIF of the group’s movement.

Place some nodes¶

You can place nodes in your simulation using several methods, which are described in the following.

Legacy positioning methods¶

Any legacy ns-3 positioning method can be used to place node in the simulation. The important additional step is to For example, you can place nodes manually like this:

Ptr<ConstantPositionMobilityModel> mm0 = enbNodes.Get (0)->GetObject<ConstantPositionMobilityModel> ();
Ptr<ConstantPositionMobilityModel> mm1 = enbNodes.Get (1)->GetObject<ConstantPositionMobilityModel> ();
mm0->SetPosition (Vector (5.0, 5.0, 1.5));
mm1->SetPosition (Vector (30.0, 40.0, 1.5));
MobilityHelper mobility;
mobility.SetMobilityModel ("ns3::ConstantPositionMobilityModel");
ueNodes.Create (2);
mobility.Install (ueNodes);
BuildingsHelper::Install (ueNodes);
mm0->SetPosition (Vector (5.0, 5.0, 1.5));
mm1->SetPosition (Vector (30.0, 40.0, 1.5));

Alternatively, you could use any existing PositionAllocator class. The coordinates of the node will determine whether it is placed outdoor or indoor and, if indoor, in which building and room it is placed.

Building-specific positioning methods¶

The following position allocator classes are available to place node in special positions with respect to buildings:

Making the Mobility Model Consistent for a node¶

Initially, a mobility model of a node is made consistent when a node is initialized, which eventually triggers a call to the DoInitialize method of the MobilityBuildingInfo` class. In particular, it calls the MakeMobilityModelConsistent method, which goes through the lists of all buildings, determine if the node is indoor or outdoor, and if indoor it also determines the building in which the node is located and the corresponding floor number inside the building. Moreover, this method also caches the position of the node, which is used to make the mobility model consistent for a moving node whenever the IsInside method of MobilityBuildingInfo class is called.

Building-aware pathloss model¶

After you placed buildings and nodes in a simulation, you can use a building-aware pathloss model in a simulation exactly in the same way you would use any regular path loss model. How to do this is specific for the wireless module that you are considering (lte, wifi, wimax, etc.), so please refer to the documentation of that model for specific instructions.

Building-aware channel condition models¶

The class BuildingsChannelConditionModel implements a channel condition model which determines the LOS/NLOS channel state based on the buildings deployed in the scenario.

The classes ThreeGppV2vUrbanChannelConditionModel and ThreeGppV2vHighwayChannelConditionModel implement hybrid channel condition models, specifically designed to model vehicular environments. More information can be found in the documentation of the propagation module.

Main configurable attributes¶

The Building class has the following configurable parameters:

The BuildingMobilityLossModel parameter configurable with the ns3 attribute system is represented by the bound (string Bounds) of the simulation area by providing a Box class with the area bounds. Moreover, by means of its methods the following parameters can be configured:

The BuildingPropagationLossModel class has the following configurable parameters configurable with the attribute system:

In order to use the hybrid mode, the class to be used is the HybridBuildingMobilityLossModel, which allows the selection of the proper pathloss model according to the pathloss logic presented in the design chapter. However, this solution has the problem that the pathloss model switching points might present discontinuities due to the different characteristics of the model. This implies that according to the specific scenario, the threshold used for switching have to be properly tuned. The simple OhBuildingMobilityLossModel overcome this problem by using only the Okumura Hata model and the wall penetration losses.

Testing Documentation¶

Overview¶

To test and validate the ns-3 Building Pathloss module, some test suites is provided which are integrated with the ns-3 test framework. To run them, you need to have configured the build of the simulator in this way:

$ ./waf configure --enable-tests --enable-modules=buildings
$ ./test.py

The above will run not only the test suites belonging to the buildings module, but also those belonging to all the other ns-3 modules on which the buildings module depends. See the ns-3 manual for generic information on the testing framework.

You can get a more detailed report in HTML format in this way:

$ ./test.py -w results.html

After the above command has run, you can view the detailed result for each test by opening the file results.html with a web browser.

You can run each test suite separately using this command:

$ ./test.py -s test-suite-name

For more details about test.py and the ns-3 testing framework, please refer to the ns-3 manual.

Description of the test suites¶

BuildingsHelper test¶

The test suite buildings-helper checks that the method BuildingsHelper::MakeAllInstancesConsistent () works properly, i.e., that the BuildingsHelper is successful in locating if nodes are outdoor or indoor, and if indoor that they are located in the correct building, room and floor. Several test cases are provided with different buildings (having different size, position, rooms and floors) and different node positions. The test passes if each every node is located correctly.

BuildingPositionAllocator test¶

The test suite building-position-allocator feature two test cases that check that respectively RandomRoomPositionAllocator and SameRoomPositionAllocator work properly. Each test cases involves a single 2x3x2 room building (total 12 rooms) at known coordinates and respectively 24 and 48 nodes. Both tests check that the number of nodes allocated in each room is the expected one and that the position of the nodes is also correct.

Buildings Pathloss tests¶

The test suite buildings-pathloss-model provides different unit tests that compare the expected results of the buildings pathloss module in specific scenarios with pre calculated values obtained offline with an Octave script (test/reference/buildings-pathloss.m). The tests are considered passed if the two values are equal up to a tolerance of 0.1, which is deemed appropriate for the typical usage of pathloss values (which are in dB).

In the following we detailed the scenarios considered, their selection has been done for covering the wide set of possible pathloss logic combinations. The pathloss logic results therefore implicitly tested.

Test #1 Okumura Hata¶

In this test we test the standard Okumura Hata model; therefore both eNB and UE are placed outside at a distance of 2000 m. The frequency used is the E-UTRA band #5, which correspond to 869 MHz (see table 5.5-1 of 36.101). The test includes also the validation of the areas extensions (i.e., urban, suburban and open-areas) and of the city size (small, medium and large).

Test #2 COST231 Model¶

This test is aimed at validating the COST231 model. The test is similar to the Okumura Hata one, except that the frequency used is the EUTRA band #1 (2140 MHz) and that the test can be performed only for large and small cities in urban scenarios due to model limitations.

Test #3 2.6 GHz model¶

This test validates the 2.6 GHz Kun model. The test is similar to Okumura Hata one except that the frequency is the EUTRA band #7 (2620 MHz) and the test can be performed only in urban scenario.

Test #4 ITU1411 LoS model¶

This test is aimed at validating the ITU1411 model in case of line of sight within street canyons transmissions. In this case the UE is placed at 100 meters far from the eNB, since the threshold for switching between LoS and NLoS is left to default one (i.e., 200 m.).

Test #5 ITU1411 NLoS model¶

This test is aimed at validating the ITU1411 model in case of non line of sight over the rooftop transmissions. In this case the UE is placed at 900 meters far from the eNB, in order to be above the threshold for switching between LoS and NLoS is left to default one (i.e., 200 m.).

Test #6 ITUP1238 model¶

This test is aimed at validating the ITUP1238 model in case of indoor transmissions. In this case both the UE and the eNB are placed in a residential building with walls made of concrete with windows. Ue is placed at the second floor and distances 30 meters far from the eNB, which is placed at the first floor.

Test #7 Outdoor -> Indoor with Okumura Hata model¶

This test validates the outdoor to indoor transmissions for large distances. In this case the UE is placed in a residential building with wall made of concrete with windows and distances 2000 meters from the outdoor eNB.

Test #8 Outdoor -> Indoor with ITU1411 model¶

This test validates the outdoor to indoor transmissions for short distances. In this case the UE is placed in a residential building with walls made of concrete with windows and distances 100 meters from the outdoor eNB.

Test #9 Indoor -> Outdoor with ITU1411 model¶

This test validates the outdoor to indoor transmissions for very short distances. In this case the eNB is placed in the second floor of a residential building with walls made of concrete with windows and distances 100 meters from the outdoor UE (i.e., LoS communication). Therefore the height gain has to be included in the pathloss evaluation.

Test #10 Indoor -> Outdoor with ITU1411 model¶

This test validates the outdoor to indoor transmissions for short distances. In this case the eNB is placed in the second floor of a residential building with walls made of concrete with windows and distances 500 meters from the outdoor UE (i.e., NLoS communication). Therefore the height gain has to be included in the pathloss evaluation.

Buildings Shadowing Test¶

The test suite buildings-shadowing-test is a unit test intended to verify the statistical distribution of the shadowing model implemented by BuildingsPathlossModel. The shadowing is modeled according to a normal distribution with mean \mu = 0 and variable standard deviation \sigma, according to models commonly used in literature. Three test cases are provided, which cover the cases of indoor, outdoor and indoor-to-outdoor communications. Each test case generates 1000 different samples of shadowing for different pairs of MobilityModel instances in a given scenario. Shadowing values are obtained by subtracting from the total loss value returned by HybridBuildingsPathlossModel the path loss component which is constant and pre-determined for each test case. The test verifies that the sample mean and sample variance of the shadowing values fall within the 99% confidence interval of the sample mean and sample variance. The test also verifies that the shadowing values returned at successive times for the same pair of MobilityModel instances is constant.

Buildings Channel Condition Model Test¶

The BuildingsChannelConditionModelTestSuite tests the class BuildingsChannelConditionModel. It checks if the channel condition between two nodes is correctly determined when a building is deployed.

References¶

[turkmani]

Turkmani A.M.D., J.D. Parson and D.G. Lewis, “Radio propagation into buildings at 441, 900 and 1400 MHz”, in Proc. of 4th Int. Conference on Land Mobile Radio, 1987.

Model Description¶

The source code for the Click model lives in the directory src/click.

Design¶

ns-3’s design is well suited for an integration with Click due to the following reasons:

The design goal was to make the ns-3-click public API simple enough such that the user needs to merely add an Ipv4ClickRouting instance to the node, and inform each Click node of the Click configuration file (.click file) that it is to use.

This model implements the interface to the Click Modular Router and provides the Ipv4ClickRouting class to allow a node to use Click for external routing. Unlike normal Ipv4RoutingProtocol sub types, Ipv4ClickRouting doesn’t use a RouteInput() method, but instead, receives a packet on the appropriate interface and processes it accordingly. Note that you need to have a routing table type element in your Click graph to use Click for external routing. This is needed by the RouteOutput() function inherited from Ipv4RoutingProtocol. Furthermore, a Click based node uses a different kind of L3 in the form of Ipv4L3ClickProtocol, which is a trimmed down version of Ipv4L3Protocol. Ipv4L3ClickProtocol passes on packets passing through the stack to Ipv4ClickRouting for processing.

Developing a Simulator API to allow ns-3 to interact with Click¶

Much of the API is already well defined, which allows Click to probe for information from the simulator (like a Node’s ID, an Interface ID and so forth). By retaining most of the methods, it should be possible to write new implementations specific to ns-3 for the same functionality.

Hence, for the Click integration with ns-3, a class named Ipv4ClickRouting will handle the interaction with Click. The code for the same can be found in src/click/model/ipv4-click-routing.{cc,h}.

Packet hand off between ns-3 and Click¶

There are four kinds of packet hand-offs that can occur between ns-3 and Click.

To overcome this, we implement Ipv4L3ClickProtocol, a stripped down version of Ipv4L3Protocol. Ipv4L3ClickProtocol passes packets to and from Ipv4ClickRouting appropriately to perform routing.

Scope and Limitations¶

References¶

Usage¶

Building Click¶

The first step is to clone Click from the github repository and build it:

$ git clone https://github.com/kohler/click
$ cd click/
$ ./configure --disable-linuxmodule --enable-nsclick --enable-wifi
$ make

The –enable-wifi flag may be skipped if you don’t intend on using Click with Wifi. * Note: You don’t need to do a ‘make install’.

Once Click has been built successfully, change into the ns-3 directory and configure ns-3 with Click Integration support:

$ ./waf configure --enable-examples --enable-tests --with-nsclick=/path/to/click/source

Hint: If you have click installed one directory above ns-3 (such as in the ns-3-allinone directory), and the name of the directory is ‘click’ (or a symbolic link to the directory is named ‘click’), then the –with-nsclick specifier is not necessary; the ns-3 build system will successfully find the directory.

If it says ‘enabled’ beside ‘NS-3 Click Integration Support’, then you’re good to go. Note: If running modular ns-3, the minimum set of modules required to run all ns-3-click examples is wifi, csma and config-store.

Next, try running one of the examples:

$ ./waf --run nsclick-simple-lan

You may then view the resulting .pcap traces, which are named nsclick-simple-lan-0-0.pcap and nsclick-simple-lan-0-1.pcap.

Click Graph Instructions¶

The following should be kept in mind when making your Click graph:

Debugging Packet Flows from Click¶

From any point within a Click graph, you may use the Print (http://read.cs.ucla.edu/click/elements/print) element and its variants for pretty printing of packet contents. Furthermore, you may generate pcap traces of packets flowing through a Click graph by using the ToDump (http://read.cs.ucla.edu/click/elements/todump) element as well. For instance:

myarpquerier


 -> Print(fromarpquery,64)


 -> ToDump(out_arpquery,PER_NODE 1)


 -> ethout;

and …will print the contents of packets that flow out of the ArpQuerier, then generate a pcap trace file which will have a suffix ‘out_arpquery’, for each node using the Click file, before pushing packets onto ‘ethout’.

Helper¶

To have a node run Click, the easiest way would be to use the ClickInternetStackHelper class in your simulation script. For instance:

ClickInternetStackHelper click;
click.SetClickFile (myNodeContainer, "nsclick-simple-lan.click");
click.SetRoutingTableElement (myNodeContainer, "u/rt");
click.Install (myNodeContainer);

The example scripts inside src/click/examples/ demonstrate the use of Click based nodes in different scenarios. The helper source can be found inside src/click/helper/click-internet-stack-helper.{h,cc}

Examples¶

The following examples have been written, which can be found in src/click/examples/:

Scripts are available within <click-dir>/conf/ that allow you to generate Click files for some common scenarios. The IP Router used in nsclick-routing.cc was generated from the make-ip-conf.pl file and slightly adapted to work with ns-3-click.

Validation¶

This model has been tested as follows:

Overview of the CSMA model¶

The ns-3 CSMA device models a simple bus network in the spirit of Ethernet. Although it does not model any real physical network you could ever build or buy, it does provide some very useful functionality.

Typically when one thinks of a bus network Ethernet or IEEE 802.3 comes to mind. Ethernet uses CSMA/CD (Carrier Sense Multiple Access with Collision Detection with exponentially increasing backoff to contend for the shared transmission medium. The ns-3 CSMA device models only a portion of this process, using the nature of the globally available channel to provide instantaneous (faster than light) carrier sense and priority-based collision “avoidance.” Collisions in the sense of Ethernet never happen and so the ns-3 CSMA device does not model collision detection, nor will any transmission in progress be “jammed.”

CSMA Layer Model¶

There are a number of conventions in use for describing layered communications architectures in the literature and in textbooks. The most common layering model is the ISO seven layer reference model. In this view the CsmaNetDevice and CsmaChannel pair occupies the lowest two layers – at the physical (layer one), and data link (layer two) positions. Another important reference model is that specified by RFC 1122, “Requirements for Internet Hosts – Communication Layers.” In this view the CsmaNetDevice and CsmaChannel pair occupies the lowest layer – the link layer. There is also a seemingly endless litany of alternative descriptions found in textbooks and in the literature. We adopt the naming conventions used in the IEEE 802 standards which speak of LLC, MAC, MII and PHY layering. These acronyms are defined as:

In this case the LLC and MAC are sublayers of the OSI data link layer and the MII and PHY are sublayers of the OSI physical layer.

The “top” of the CSMA device defines the transition from the network layer to the data link layer. This transition is performed by higher layers by calling either CsmaNetDevice::Send or CsmaNetDevice::SendFrom.

In contrast to the IEEE 802.3 standards, there is no precisely specified PHY in the CSMA model in the sense of wire types, signals or pinouts. The “bottom” interface of the CsmaNetDevice can be thought of as as a kind of Media Independent Interface (MII) as seen in the “Fast Ethernet” (IEEE 802.3u) specifications. This MII interface fits into a corresponding media independent interface on the CsmaChannel. You will not find the equivalent of a 10BASE-T or a 1000BASE-LX PHY.

The CsmaNetDevice calls the CsmaChannel through a media independent interface. There is a method defined to tell the channel when to start “wiggling the wires” using the method CsmaChannel::TransmitStart, and a method to tell the channel when the transmission process is done and the channel should begin propagating the last bit across the “wire”: CsmaChannel::TransmitEnd.

When the TransmitEnd method is executed, the channel will model a single uniform signal propagation delay in the medium and deliver copes of the packet to each of the devices attached to the packet via the CsmaNetDevice::Receive method.

There is a “pin” in the device media independent interface corresponding to “COL” (collision). The state of the channel may be sensed by calling CsmaChannel::GetState. Each device will look at this “pin” before starting a send and will perform appropriate backoff operations if required.

Properly received packets are forwarded up to higher levels from the CsmaNetDevice via a callback mechanism. The callback function is initialized by the higher layer (when the net device is attached) using CsmaNetDevice::SetReceiveCallback and is invoked upon “proper” reception of a packet by the net device in order to forward the packet up the protocol stack.

CSMA Channel Model¶

The class CsmaChannel models the actual transmission medium. There is no fixed limit for the number of devices connected to the channel. The CsmaChannel models a data rate and a speed-of-light delay which can be accessed via the attributes “DataRate” and “Delay” respectively. The data rate provided to the channel is used to set the data rates used by the transmitter sections of the CSMA devices connected to the channel. There is no way to independently set data rates in the devices. Since the data rate is only used to calculate a delay time, there is no limitation (other than by the data type holding the value) on the speed at which CSMA channels and devices can operate; and no restriction based on any kind of PHY characteristics.

The CsmaChannel has three states, IDLE, TRANSMITTING and PROPAGATING. These three states are “seen” instantaneously by all devices on the channel. By this we mean that if one device begins or ends a simulated transmission, all devices on the channel are immediately aware of the change in state. There is no time during which one device may see an IDLE channel while another device physically further away in the collision domain may have begun transmitting with the associated signals not propagated down the channel to other devices. Thus there is no need for collision detection in the CsmaChannel model and it is not implemented in any way.

We do, as the name indicates, have a Carrier Sense aspect to the model. Since the simulator is single threaded, access to the common channel will be serialized by the simulator. This provides a deterministic mechanism for contending for the channel. The channel is allocated (transitioned from state IDLE to state TRANSMITTING) on a first-come first-served basis. The channel always goes through a three state process:

IDLE -> TRANSMITTING -> PROPAGATING -> IDLE

The TRANSMITTING state models the time during which the source net device is actually wiggling the signals on the wire. The PROPAGATING state models the time after the last bit was sent, when the signal is propagating down the wire to the “far end.”

The transition to the TRANSMITTING state is driven by a call to CsmaChannel::TransmitStart which is called by the net device that transmits the packet. It is the responsibility of that device to end the transmission with a call to CsmaChannel::TransmitEnd at the appropriate simulation time that reflects the time elapsed to put all of the packet bits on the wire. When TransmitEnd is called, the channel schedules an event corresponding to a single speed-of-light delay. This delay applies to all net devices on the channel identically. You can think of a symmetrical hub in which the packet bits propagate to a central location and then back out equal length cables to the other devices on the channel. The single “speed of light” delay then corresponds to the time it takes for: 1) a signal to propagate from one CsmaNetDevice through its cable to the hub; plus 2) the time it takes for the hub to forward the packet out a port; plus 3) the time it takes for the signal in question to propagate to the destination net device.

The CsmaChannel models a broadcast medium so the packet is delivered to all of the devices on the channel (including the source) at the end of the propagation time. It is the responsibility of the sending device to determine whether or not it receives a packet broadcast over the channel.

The CsmaChannel provides following Attributes:

CSMA Net Device Model¶

The CSMA network device appears somewhat like an Ethernet device. The CsmaNetDevice provides following Attributes:

The CsmaNetDevice supports the assignment of a “receive error model.” This is an ErrorModel object that is used to simulate data corruption on the link.

Packets sent over the CsmaNetDevice are always routed through the transmit queue to provide a trace hook for packets sent out over the network. This transmit queue can be set (via attribute) to model different queuing strategies.

Also configurable by attribute is the encapsulation method used by the device. Every packet gets an EthernetHeader that includes the destination and source MAC addresses, and a length/type field. Every packet also gets an EthernetTrailer which includes the FCS. Data in the packet may be encapsulated in different ways.

By default, or by setting the “EncapsulationMode” attribute to “Dix”, the encapsulation is according to the DEC, Intel, Xerox standard. This is sometimes called EthernetII framing and is the familiar destination MAC, source MAC, EtherType, Data, CRC format.

If the “EncapsulationMode” attribute is set to “Llc”, the encapsulation is by LLC SNAP. In this case, a SNAP header is added that contains the EtherType (IP or ARP).

The other implemented encapsulation modes are IP_ARP (set “EncapsulationMode” to “IpArp”) in which the length type of the Ethernet header receives the protocol number of the packet; or ETHERNET_V1 (set “EncapsulationMode” to “EthernetV1”) in which the length type of the Ethernet header receives the length of the packet. A “Raw” encapsulation mode is defined but not implemented – use of the RAW mode results in an assertion.

Note that all net devices on a channel must be set to the same encapsulation mode for correct results. The encapsulation mode is not sensed at the receiver.

The CsmaNetDevice implements a random exponential backoff algorithm that is executed if the channel is determined to be busy (TRANSMITTING or PPROPAGATING) when the device wants to start propagating. This results in a random delay of up to pow (2, retries) - 1 microseconds before a retry is attempted. The default maximum number of retries is 1000.

Using the CsmaNetDevice¶

The CSMA net devices and channels are typically created and configured using the associated CsmaHelper object. The various ns-3 device helpers generally work in a similar way, and their use is seen in many of our example programs.

The conceptual model of interest is that of a bare computer “husk” into which you plug net devices. The bare computers are created using a NodeContainer helper. You just ask this helper to create as many computers (we call them Nodes) as you need on your network:

NodeContainer csmaNodes;
csmaNodes.Create (nCsmaNodes);

Once you have your nodes, you need to instantiate a CsmaHelper and set any attributes you may want to change.:

CsmaHelper csma;
csma.SetChannelAttribute ("DataRate", StringValue ("100Mbps"));
csma.SetChannelAttribute ("Delay", TimeValue (NanoSeconds (6560)));
csma.SetDeviceAttribute ("EncapsulationMode", StringValue ("Dix"));
csma.SetDeviceAttribute ("FrameSize", UintegerValue (2000));

Once the attributes are set, all that remains is to create the devices and install them on the required nodes, and to connect the devices together using a CSMA channel. When we create the net devices, we add them to a container to allow you to use them in the future. This all takes just one line of code.:

NetDeviceContainer csmaDevices = csma.Install (csmaNodes);

We recommend thinking carefully about changing these Attributes, since it can result in behavior that surprises users. We allow this because we believe flexibility is important. As an example of a possibly surprising effect of changing Attributes, consider the following:

The Mtu Attribute indicates the Maximum Transmission Unit to the device. This is the size of the largest Protocol Data Unit (PDU) that the device can send. This Attribute defaults to 1500 bytes and corresponds to a number found in RFC 894, “A Standard for the Transmission of IP Datagrams over Ethernet Networks.” The number is actually derived from the maximum packet size for 10Base5 (full-spec Ethernet) networks – 1518 bytes. If you subtract DIX encapsulation overhead for Ethernet packets (18 bytes) you will end up with a maximum possible data size (MTU) of 1500 bytes. One can also find that the MTU for IEEE 802.3 networks is 1492 bytes. This is because LLC/SNAP encapsulation adds an extra eight bytes of overhead to the packet. In both cases, the underlying network hardware is limited to 1518 bytes, but the MTU is different because the encapsulation is different.

If one leaves the Mtu Attribute at 1500 bytes and changes the encapsulation mode Attribute to Llc, the result will be a network that encapsulates 1500 byte PDUs with LLC/SNAP framing resulting in packets of 1526 bytes. This would be illegal in many networks, but we allow you do do this. This results in a simulation that quite subtly does not reflect what you might be expecting since a real device would balk at sending a 1526 byte packet.

There also exist jumbo frames (1500 < MTU <= 9000 bytes) and super-jumbo (MTU > 9000 bytes) frames that are not officially sanctioned by IEEE but are available in some high-speed (Gigabit) networks and NICs. In the CSMA model, one could leave the encapsulation mode set to Dix, and set the Mtu to 64000 bytes – even though an associated CsmaChannel DataRate was left at 10 megabits per second (certainly not Gigabit Ethernet). This would essentially model an Ethernet switch made out of vampire-tapped 1980s-style 10Base5 networks that support super-jumbo datagrams, which is certainly not something that was ever made, nor is likely to ever be made; however it is quite easy for you to configure.

Be careful about assumptions regarding what CSMA is actually modelling and how configuration (Attributes) may allow you to swerve considerably away from reality.

CSMA Tracing¶

Like all ns-3 devices, the CSMA Model provides a number of trace sources. These trace sources can be hooked using your own custom trace code, or you can use our helper functions to arrange for tracing to be enabled on devices you specify.

Upper-Level (MAC) Hooks¶

From the point of view of tracing in the net device, there are several interesting points to insert trace hooks. A convention inherited from other simulators is that packets destined for transmission onto attached networks pass through a single “transmit queue” in the net device. We provide trace hooks at this point in packet flow, which corresponds (abstractly) only to a transition from the network to data link layer, and call them collectively the device MAC hooks.

When a packet is sent to the CSMA net device for transmission it always passes through the transmit queue. The transmit queue in the CsmaNetDevice inherits from Queue, and therefore inherits three trace sources:

The upper-level (MAC) trace hooks for the CsmaNetDevice are, in fact, exactly these three trace sources on the single transmit queue of the device.

The m_traceEnqueue event is triggered when a packet is placed on the transmit queue. This happens at the time that CsmaNetDevice::Send or CsmaNetDevice::SendFrom is called by a higher layer to queue a packet for transmission.

The m_traceDequeue event is triggered when a packet is removed from the transmit queue. Dequeues from the transmit queue can happen in three situations: 1) If the underlying channel is idle when the CsmaNetDevice::Send or CsmaNetDevice::SendFrom is called, a packet is dequeued from the transmit queue and immediately transmitted; 2) If the underlying channel is idle, a packet may be dequeued and immediately transmitted in an internal TransmitCompleteEvent that functions much like a transmit complete interrupt service routine; or 3) from the random exponential backoff handler if a timeout is detected.

Case (3) implies that a packet is dequeued from the transmit queue if it is unable to be transmitted according to the backoff rules. It is important to understand that this will appear as a Dequeued packet and it is easy to incorrectly assume that the packet was transmitted since it passed through the transmit queue. In fact, a packet is actually dropped by the net device in this case. The reason for this behavior is due to the definition of the Queue Drop event. The m_traceDrop event is, by definition, fired when a packet cannot be enqueued on the transmit queue because it is full. This event only fires if the queue is full and we do not overload this event to indicate that the CsmaChannel is “full.”

Lower-Level (PHY) Hooks¶

Similar to the upper level trace hooks, there are trace hooks available at the lower levels of the net device. We call these the PHY hooks. These events fire from the device methods that talk directly to the CsmaChannel.

The trace source m_dropTrace is called to indicate a packet that is dropped by the device. This happens in two cases: First, if the receive side of the net device is not enabled (see CsmaNetDevice::m_receiveEnable and the associated attribute “ReceiveEnable”).

The m_dropTrace is also used to indicate that a packet was discarded as corrupt if a receive error model is used (see CsmaNetDevice::m_receiveErrorModel and the associated attribute “ReceiveErrorModel”).

The other low-level trace source fires on reception of an accepted packet (see CsmaNetDevice::m_rxTrace). A packet is accepted if it is destined for the broadcast address, a multicast address, or to the MAC address assigned to the net device.

Summary¶

The ns3 CSMA model is a simplistic model of an Ethernet-like network. It supports a Carrier-Sense function and allows for Multiple Access to a shared medium. It is not physical in the sense that the state of the medium is instantaneously shared among all devices. This means that there is no collision detection required in this model and none is implemented. There will never be a “jam” of a packet already on the medium. Access to the shared channel is on a first-come first-served basis as determined by the simulator scheduler. If the channel is determined to be busy by looking at the global state, a random exponential backoff is performed and a retry is attempted.

Ns-3 Attributes provide a mechanism for setting various parameters in the device and channel such as addresses, encapsulation modes and error model selection. Trace hooks are provided in the usual manner with a set of upper level hooks corresponding to a transmit queue and used in ASCII tracing; and also a set of lower level hooks used in pcap tracing.

Although the ns-3 CsmaChannel and CsmaNetDevice does not model any kind of network you could build or buy, it does provide us with some useful functionality. You should, however, understand that it is explicitly not Ethernet or any flavor of IEEE 802.3 but an interesting subset.

DSDV Routing Overview¶

DSDV Routing Table: Every node will maintain a table listing all the other nodes it has known either directly or through some neighbors. Every node has a single entry in the routing table. The entry will have information about the node’s IP address, last known sequence number and the hop count to reach that node. Along with these details the table also keeps track of the nexthop neighbor to reach the destination node, the timestamp of the last update received for that node.

The DSDV update message consists of three fields, Destination Address, Sequence Number and Hop Count.

Each node uses 2 mechanisms to send out the DSDV updates. They are,

The updates are accepted based on the metric for a particular node. The first factor determining the acceptance of an update is the sequence number. It has to accept the update if the sequence number of the update message is higher irrespective of the metric. If the update with same sequence number is received, then the update with least metric (hopCount) is given precedence.

In highly mobile scenarios, there is a high chance of route fluctuations, thus we have the concept of weighted settling time where an update with change in metric will not be advertised to neighbors. The node waits for the settling time to make sure that it did not receive the update from its old neighbor before sending out that update.

The current implementation covers all the above features of DSDV. The current implementation also has a request queue to buffer packets that have no routes to destination. The default is set to buffer up to 5 packets per destination.

References¶

Link to the Paper: http://portal.acm.org/citation.cfm?doid=190314.190336

DSR Routing Overview¶

This model implements the base specification of the Dynamic Source Routing (DSR) protocol. Implementation is based on RFC 4728, with some extensions and modifications to the RFC specifications.

DSR operates on a on-demand behavior. Therefore, our DSR model buffers all packets while a route request packet (RREQ) is disseminated. We implement a packet buffer in dsr-rsendbuff.cc. The packet queue implements garbage collection of old packets and a queue size limit. When the packet is sent out from the send buffer, it will be queued in maintenance buffer for next hop acknowledgment.

The maintenance buffer then buffers the already sent out packets and waits for the notification of packet delivery. Protocol operation strongly depends on broken link detection mechanism. We implement the three heuristics recommended based the RFC as follows:

First, we use link layer feedback when possible, which is also the fastest mechanism of these three to detect link errors. A link is considered to be broken if frame transmission results in a transmission failure for all retries. This mechanism is meant for active links and works much faster than in its absence. DSR is able to detect the link layer transmission failure and notify that as broken. Recalculation of routes will be triggered when needed. If user does not want to use link layer acknowledgment, it can be tuned by setting “LinkAcknowledgment” attribute to false in “dsr-routing.cc”.

Second, passive acknowledgment should be used whenever possible. The node turns on “promiscuous” receive mode, in which it can receive packets not destined for itself, and when the node assures the delivery of that data packet to its destination, it cancels the passive acknowledgment timer.

Last, we use a network layer acknowledge scheme to notify the receipt of a packet. Route request packet will not be acknowledged or retransmitted.

The Route Cache implementation support garbage collection of old entries and state machine, as defined in the standard. It implements as a STL map container. The key is the destination IP address.

DSR operates with direct access to IP header, and operates between network and transport layer. When packet is sent out from transport layer, it passes itself to DSR and DSR header is appended.

We have two caching mechanisms: path cache and link cache. The path cache saves the whole path in the cache. The paths are sorted based on the hop count, and whenever one path is not able to be used, we change to the next path. The link cache is a slightly better design in the sense that it uses different subpaths and uses Implemented Link Cache using Dijkstra algorithm, and this part is implemented by Song Luan <lsuper@mail.ustc.edu.cn>.

The following optional protocol optimizations aren’t implemented:

DSR update in ns-3.17¶

We originally used “TxErrHeader” in Ptr<WifiMac> to indicate the transmission error of a specific packet in link layer, however, it was not working quite correctly since even when the packet was dropped, this header was not recorded in the trace file. We used to a different path on implementing the link layer notification mechanism. We look into the trace file by finding packet receive event. If we find one receive event for the data packet, we count that as the indicator for successful data delivery.

Useful parameters¶

+------------------------- +------------------------------------+-------------+
| Parameter                | Description                        | Default     |
+==========================+====================================+=============+
| MaxSendBuffLen           | Maximum number of packets that can | 64          |
|                          | be stored in send buffer           |             |
+------------------------- +------------------------------------+-------------+
| MaxSendBuffTime          | Maximum time packets can be queued | Seconds(30) |
|                          | in the send buffer                 |             |
+------------------------- +------------------------------------+-------------+
| MaxMaintLen              | Maximum number of packets that can | 50          |
|                          | be stored in maintenance buffer    |             |
+------------------------- +------------------------------------+-------------+
| MaxMaintTime             | Maximum time packets can be queued | Seconds(30) |
|                          | in maintenance buffer              |             |
+------------------------- +------------------------------------+-------------+
| MaxCacheLen              | Maximum number of route entries    | 64          |
|                          | that can be stored in route cache  |             |
+------------------------- +------------------------------------+-------------+
| RouteCacheTimeout        | Maximum time the route cache can   | Seconds(300)|
|                          | be queued in route cache           |             |
+------------------------- +------------------------------------+-------------+
| RreqRetries              | Maximum number of retransmissions  | 16          |
|                          | for request discovery of a route   |             |
+------------------------- +------------------------------------+-------------+
| CacheType                | Use Link Cache or use Path Cache   | "LinkCache" |
|                          |                                    |             |
+------------------------- +------------------------------------+-------------+
| LinkAcknowledgment       | Enable Link layer acknowledgment   | True        |
|                          | mechanism                          |             |
+------------------------- +------------------------------------+-------------+

Implementation modification¶

Current Route Cache implementation¶

This implementation used “path cache”, which is simple to implement and ensures loop-free paths:

DSR Instructions¶

The following should be kept in mind when running DSR as routing protocol:

Helper¶

To have a node run DSR, the easiest way would be to use the DsrHelper and DsrMainHelpers in your simulation script. For instance:

DsrHelper dsr;
DsrMainHelper dsrMain;
dsrMain.Install (dsr, adhocNodes);

The example scripts inside src/dsr/examples/ demonstrate the use of DSR based nodes in different scenarios. The helper source can be found inside src/dsr/helper/dsr-main-helper.{h,cc} and src/dsr/helper/dsr-helper.{h,cc}

Examples¶

The example can be found in src/dsr/examples/:

DSR is also built in the routing comparison case in examples/routing/:

Validation¶

This model has been tested as follows:

A paper was presented on these results at the Workshop on ns-3 in 2011.

Limitations¶

The model is not fully compliant with RFC 4728. As an example, Dsr fixed size header has been extended and it is four octets longer then the RFC specification. As a consequence, the DSR headers can not be correctly decoded by Wireshark.

The model full compliance with the RFC is planned for the future.

References¶

[1] Original paper: http://www.monarch.cs.rice.edu/monarch-papers/dsr-chapter00.pdf

[2] RFC 4728 http://www6.ietf.org/rfc/rfc4728.txt

[3] Broch’s comparison paper: http://www.monarch.cs.rice.edu/monarch-papers/mobicom98.ps

File Descriptor NetDevice¶

The src/fd-net-device module provides the FdNetDevice class, which is able to read and write traffic using a file descriptor provided by the user. This file descriptor can be associated to a TAP device, to a raw socket, to a user space process generating/consuming traffic, etc. The user has full freedom to define how external traffic is generated and ns-3 traffic is consumed.

Different mechanisms to associate a simulation to external traffic can be provided through helper classes. Three specific helpers are provided:

Model Description¶

The source code for this module lives in the directory src/fd-net-device.

The FdNetDevice is a special type of ns-3 NetDevice that reads traffic to and from a file descriptor. That is, unlike pure simulation NetDevice objects that write frames to and from a simulated channel, this FdNetDevice directs frames out of the simulation to a file descriptor. The file descriptor may be associated to a Linux TUN/TAP device, to a socket, or to a user-space process.

It is up to the user of this device to provide a file descriptor. The type of file descriptor being provided determines what is being modelled. For instance, if the file descriptor provides a raw socket to a WiFi card on the host machine, the device being modelled is a WiFi device.

From the conceptual “top” of the device looking down, it looks to the simulated node like a device supporting a 48-bit IEEE MAC address that can be bridged, supports broadcast, and uses IPv4 ARP or IPv6 Neighbor Discovery, although these attributes can be tuned on a per-use-case basis.

Design¶

The FdNetDevice implementation makes use of a reader object, extended from the FdReader class in the ns-3 src/core module, which manages a separate thread from the main ns-3 execution thread, in order to read traffic from the file descriptor.

Upon invocation of the StartDevice method, the reader object is initialized and starts the reading thread. Before device start, a file descriptor must be previously associated to the FdNetDevice with the SetFileDescriptor invocation.

The creation and configuration of the file descriptor can be left to a number of helpers, described in more detail below. When this is done, the invocation of SetFileDescriptor is responsibility of the helper and must not be directly invoked by the user.

Upon reading an incoming frame from the file descriptor, the reader will pass the frame to the ReceiveCallback method, whose task it is to schedule the reception of the frame by the device as a ns-3 simulation event. Since the new frame is passed from the reader thread to the main ns-3 simulation thread, thread-safety issues are avoided by using the ScheduleWithContext call instead of the regular Schedule call.

In order to avoid overwhelming the scheduler when the incoming data rate is too high, a counter is kept with the number of frames that are currently scheduled to be received by the device. If this counter reaches the value given by the RxQueueSize attribute in the device, then the new frame will be dropped silently.

The actual reception of the new frame by the device occurs when the scheduled FordwarUp method is invoked by the simulator. This method acts as if a new frame had arrived from a channel attached to the device. The device then decapsulates the frame, removing any layer 2 headers, and forwards it to upper network stack layers of the node. The ForwardUp method will remove the frame headers, according to the frame encapsulation type defined by the EncapsulationMode attribute, and invoke the receive callback passing an IP packet.

An extra header, the PI header, can be present when the file descriptor is associated to a TAP device that was created without setting the IFF_NO_PI flag. This extra header is removed if EncapsulationMode is set to DIXPI value.

In the opposite direction, packets generated inside the simulation that are sent out through the device, will be passed to the Send method, which will in turn invoke the SendFrom method. The latter method will add the necessary layer 2 headers, and simply write the newly created frame to the file descriptor.

Scope and Limitations¶

Users of this device are cautioned that there is no flow control across the file descriptor boundary, when using in emulation mode. That is, in a Linux system, if the speed of writing network packets exceeds the ability of the underlying physical device to buffer the packets, backpressure up to the writing application will be applied to avoid local packet loss. No such flow control is provided across the file descriptor interface, so users must be aware of this limitation.

As explained before, the RxQueueSize attribute limits the number of packets that can be pending to be received by the device. Frames read from the file descriptor while the number of pending packets is in its maximum will be silently dropped.

The mtu of the device defaults to the Ethernet II MTU value. However, helpers are supposed to set the mtu to the right value to reflect the characteristics of the network interface associated to the file descriptor. If no helper is used, then the responsibility of setting the correct mtu value for the device falls back to the user. The size of the read buffer on the file descriptor reader is set to the mtu value in the StartDevice method.

The FdNetDevice class currently supports three encapsulation modes, DIX for Ethernet II frames, LLC for 802.2 LLC/SNAP frames, and DIXPI for Ethernet II frames with an additional TAP PI header. This means that traffic traversing the file descriptor is expected to be Ethernet II compatible. IEEE 802.1q (VLAN) tagging is not supported. Attaching an FdNetDevice to a wireless interface is possible as long as the driver provides Ethernet II frames to the socket API. Note that to associate a FdNetDevice to a wireless card in ad-hoc mode, the MAC address of the device must be set to the real card MAC address, else any incoming traffic a fake MAC address will be discarded by the driver.

As mentioned before, three helpers are provided with the fd-net-device module. Each individual helper (file descriptor type) may have platform limitations. For instance, threading, real-time simulation mode, and the ability to create TUN/TAP devices are prerequisites to using the provided helpers. Support for these modes can be found in the output of the waf configure step, e.g.:

Threading Primitives          : enabled
Real Time Simulator           : enabled
Emulated Net Device           : enabled
Tap Bridge                    : enabled

It is important to mention that while testing the FdNetDevice we have found an upper bound limit for TCP throughput when using 1Gb Ethernet links of 60Mbps. This limit is most likely due to the processing power of the computers involved in the tests.

Usage¶

The usage pattern for this type of device is similar to other net devices with helpers that install to node pointers or node containers. When using the base FdNetDeviceHelper the user is responsible for creating and setting the file descriptor by himself.

FdNetDeviceHelper fd;
NetDeviceContainer devices = fd.Install (nodes);
// file descriptor generation
...
device->SetFileDescriptor (fd);

Most commonly a FdNetDevice will be used to interact with the host system. In these cases it is almost certain that the user will want to run in real-time emulation mode, and to enable checksum computations. The typical program statements are as follows:

GlobalValue::Bind ("SimulatorImplementationType", StringValue ("ns3::RealtimeSimulatorImpl"));
GlobalValue::Bind ("ChecksumEnabled", BooleanValue (true));

The easiest way to set up an experiment that interacts with a Linux host system is to user the Emu and Tap helpers. Perhaps the most unusual part of these helper implementations relates to the requirement for executing some of the code with super-user permissions. Rather than force the user to execute the entire simulation as root, we provide a small “creator” program that runs as root and does any required high-permission sockets work. The easiest way to set the right privileges for the “creator” programs, is by enabling the --enable-sudo flag when performing waf configure.

We do a similar thing for both the Emu and the Tap devices. The high-level view is that the CreateFileDescriptor method creates a local interprocess (Unix) socket, forks, and executes the small creation program. The small program, which runs as suid root, creates a raw socket and sends back the raw socket file descriptor over the Unix socket that is passed to it as a parameter. The raw socket is passed as a control message (sometimes called ancillary data) of type SCM_RIGHTS.

Helpers¶

EmuFdNetDeviceHelper¶

The EmuFdNetDeviceHelper creates a raw socket to an underlying physical device, and provides the socket descriptor to the FdNetDevice. This allows the ns-3 simulation to read frames from and write frames to a network device on the host.

The emulation helper permits to transparently integrate a simulated ns-3 node into a network composed of real nodes.

+----------------------+     +-----------------------+
|         host 1       |     |         host 2        |
+----------------------+     +-----------------------+
|    ns-3 simulation   |     |                       |
+----------------------+     |         Linux         |
|       ns-3 Node      |     |     Network Stack     |
|  +----------------+  |     |   +----------------+  |
|  |    ns-3 TCP    |  |     |   |       TCP      |  |
|  +----------------+  |     |   +----------------+  |
|  |    ns-3 IP     |  |     |   |       IP       |  |
|  +----------------+  |     |   +----------------+  |
|  |   FdNetDevice  |  |     |   |                |  |
|  |    10.1.1.1    |  |     |   |                |  |
|  +----------------+  |     |   +    ETHERNET    +  |
|  |   raw socket   |  |     |   |                |  |
|--+----------------+--|     |   +----------------+  |
|       | eth0 |       |     |        | eth0 |       |
+-------+------+-------+     +--------+------+-------+


        10.1.1.11                     10.1.1.12


            |                            |


            +----------------------------+

This helper replaces the functionality of the EmuNetDevice found in ns-3 prior to ns-3.17, by bringing this type of device into the common framework of the FdNetDevice. The EmuNetDevice was deprecated in favor of this new helper.

The device is configured to perform MAC spoofing to separate simulation network traffic from other network traffic that may be flowing to and from the host.

One can use this helper in a testbed situation where the host on which the simulation is running has a specific interface of interest which drives the testbed hardware. You would also need to set this specific interface into promiscuous mode and provide an appropriate device name to the ns-3 simulation. Additionally, hardware offloading of segmentation and checksums should be disabled.

The helper only works if the underlying interface is up and in promiscuous mode. Packets will be sent out over the device, but we use MAC spoofing. The MAC addresses will be generated (by default) using the Organizationally Unique Identifier (OUI) 00:00:00 as a base. This vendor code is not assigned to any organization and so should not conflict with any real hardware.

It is always up to the user to determine that using these MAC addresses is okay on your network and won’t conflict with anything else (including another simulation using such devices) on your network. If you are using the emulated FdNetDevice configuration in separate simulations, you must consider global MAC address assignment issues and ensure that MAC addresses are unique across all simulations. The emulated net device respects the MAC address provided in the Address attribute so you can do this manually. For larger simulations, you may want to set the OUI in the MAC address allocation function.

Before invoking the Install method, the correct device name must be configured on the helper using the SetDeviceName method. The device name is required to identify which physical device should be used to open the raw socket.

EmuFdNetDeviceHelper emu;
emu.SetDeviceName (deviceName);
NetDeviceContainer devices = emu.Install (node);
Ptr<NetDevice> device = devices.Get (0);
device->SetAttribute ("Address", Mac48AddressValue (Mac48Address::Allocate ()));

TapFdNetDeviceHelper¶

A Tap device is a special type of Linux device for which one end of the device appears to the kernel as a virtual net_device, and the other end is provided as a file descriptor to user-space. This file descriptor can be passed to the FdNetDevice. Packets forwarded to the TAP device by the kernel will show up in the FdNetDevice in ns-3.

Users should note that this usage of TAP devices is different than that provided by the TapBridge NetDevice found in src/tap-bridge. The model in this helper is as follows:

+-------------------------------------+
|                host                 |
+-------------------------------------+
|    ns-3 simulation   |              |
+----------------------+              |
|      ns-3 Node       |              |
|  +----------------+  |              |
|  |    ns-3 TCP    |  |              |
|  +----------------+  |              |
|  |    ns-3 IP     |  |              |
|  +----------------+  |              |
|  |   FdNetDevice  |  |              |
|--+----------------+--+    +------+  |
|       | TAP  |            | eth0 |  |
|       +------+            +------+  |
|     192.168.0.1               |     |
+-------------------------------|-----+


                                |


                                |


                                ------------ (Internet) -----

In the above, the configuration requires that the host be able to forward traffic generated by the simulation to the Internet.

The model in TapBridge (in another module) is as follows:

+--------+
|  Linux |
|  host  |                    +----------+
| ------ |                    |   ghost  |
|  apps  |                    |   node   |
| ------ |                    | -------- |
|  stack |                    |    IP    |     +----------+
| ------ |                    |   stack  |     |   node   |
|  TAP   |                    |==========|     | -------- |
| device | <----- IPC ------> |   tap    |     |    IP    |
+--------+                    |  bridge  |     |   stack  |


                              | -------- |     | -------- |


                              |   ns-3   |     |   ns-3   |


                              |   net    |     |   net    |


                              |  device  |     |  device  |


                              +----------+     +----------+


                                   ||               ||


                              +---------------------------+


                              |        ns-3 channel       |


                              +---------------------------+

In the above, packets instead traverse ns-3 NetDevices and Channels.

The usage pattern for this example is that the user sets the MAC address and either (or both) the IPv4 and IPv6 addresses and masks on the device, and the PI header if needed. For example:

TapFdNetDeviceHelper helper;
helper.SetDeviceName (deviceName);
helper.SetModePi (modePi);
helper.SetTapIpv4Address (tapIp);
helper.SetTapIpv4Mask (tapMask);
...
helper.Install (node);

PlanetLabFdNetDeviceHelper¶

PlanetLab is a world wide distributed network testbed composed of nodes connected to the Internet. Running ns-3 simulations in PlanetLab nodes using the PlanetLabFdNetDeviceHelper allows to send simulated traffic generated by ns-3 directly to the Internet. This setup can be useful to validate ns-3 Internet protocols or other future protocols implemented in ns-3.

To run experiments using PlanetLab nodes it is required to have a PlanetLab account. Only members of PlanetLab partner institutions can obtain such accounts ( for more information visit http://www.planet-lab.org/ or http://www.planet-lab.eu ). Once the account is obtained, a PlanetLab slice must be requested in order to conduct experiments. A slice represents an experiment unit related to a group of PlanetLab users, and can be associated to virtual machines in different PlanetLab nodes. Slices can also be customized by adding configuration tags to it (this is done by PlanetLab administrators).

The PlanetLabFdNetDeviceHelper creates TAP devices on PlanetLab nodes using specific PlanetLab mechanisms (i.e. the vsys system), and associates the TAP device to a FdNetDevice in ns-3. The functionality provided by this helper is similar to that provided by the FdTapNetDeviceHelper, except that the underlying mechanisms to create the TAP device are different.

+-------------------------------------+
|         PlanetLab  host             |
+-------------------------------------+
|    ns-3 simulation   |              |
+----------------------+              |
|       ns-3 Node      |              |
|  +----------------+  |              |
|  |    ns-3 TCP    |  |              |
|  +----------------+  |              |
|  |    ns-3 IP     |  |              |
|  +----------------+  |              |
|  |   FdNetDevice  |  |              |
|--+----------------+--+    +------+  |
|       | TAP  |            | eth0 |  |
|       +------+            +------+  |
|     192.168.0.1               |     |
+-------------------------------|-----+


                                |


                                |


                                ------------ (Internet) -----

In order to be able to assign private IPv4 addresses to the TAP devices, account holders must request the vsys_vnet tag to be added to their slice by PlanetLab administrators. The vsys_vnet tag is associated to private network segment and only addresses from this segment can be used in experiments.

The syntax used to create a TAP device with this helper is similar to that used for the previously described helpers:

PlanetLabFdNetDeviceHelper helper;
helper.SetTapIpAddress (tapIp);
helper.SetTapMask (tapMask);
...
helper.Install (node);

PlanetLab nodes have a Fedora based distribution, so ns-3 can be installed following the instructions for ns-3 Linux installation.

Attributes¶

The FdNetDevice provides a number of attributes:

Start and Stop do not normally need to be specified unless the user wants to limit the time during which this device is active. Address needs to be set to some kind of unique MAC address if the simulation will be interacting with other real devices somehow using real MAC addresses. Typical code:

device->SetAttribute ("Address", Mac48AddressValue (Mac48Address::Allocate ()));

Output¶

Ascii and PCAP tracing is provided similar to the other ns-3 NetDevice types, through the helpers, such as (e.g.):

The standard set of Mac-level NetDevice trace sources is provided.

Examples¶

Several examples are provided:

Netmap NetDevice¶

The fd-net-device module provides the NetmapNetDevice class, a class derived from the FdNetDevice which is able to read and write traffic using a netmap file descriptor. This netmap file descriptor must be associated to a real ethernet device in the host machine. The NetmapNetDeviceHelper class supports the configuration of a NetmapNetDevice.

netmap is a fast packet processing capability that bypasses the host networking stack and gains direct access to network device. netmap was developed by Luigi Rizzo [Rizzo2012] and is maintained as an open source project on GitHub at https://github.com/luigirizzo/netmap.

The NetmapNetDevice for ns-3 [Imputato2019] was developed by Pasquale Imputato in the 2017-19 timeframe. The use of NetmapNetDevice requires that the host system has netmap support (and for best performance, the drivers must support netmap and must be using a netmap-enabled device driver). Users can expect that emulation support using Netmap will support higher packets per second than emulation using FdNetDevice with raw sockets (which pass through the Linux networking kernel).

[Rizzo2012]

Luigi Rizzo, “netmap: A Novel Framework for Fast Packet I/O”, Proceedings of 2012 USENIX Annual Techincal Conference, June 2012.

[Imputato2019]

Pasquale Imputato, Stefano Avallone, Enhancing the fidelity of network emulation through direct access to device buffers, Journal of Network and Computer Applications, Volume 130, 2019, Pages 63-75, (http://www.sciencedirect.com/science/article/pii/S1084804519300220)

Model Description¶

Design¶

Because netmap uses file descriptor based communication to interact with the real device, the straightforward approach to design a new NetDevice around netmap is to have it inherit from the existing FdNetDevice and implement a specialized version of the operations specific to netmap. The operations that require a specialized implementation are the initialization, because the NIC has to be put in netmap mode, and the read/write methods, which have to make use of the netmap API to coordinate the exchange of packets with the netmap rings.

In the initialization stage, the network device is switched to netmap mode, so that ns-3 is able to send/receive packets to/from the real network device by writing/reading them to/from the netmap rings. Following the design of the FdNetDevice, a separate reading thread is started during the initialization. The task of the reading thread is to wait for new incoming packets in the netmap receiver rings, in order to schedule the events of packet reception. In the initialization of the NetmapNetDevice, an additional thread, the sync thread, is started. The sync thread is required because, in order to reduce the cost of the system calls, netmap does not automatically transfer a packet written to a slot of the netmap ring to the transmission ring or to the installed qdisc. It is up to the user process to periodically request a synchronization of the netmap ring. Therefore, the purpose of the sync thread is to periodically make a TXSYNC ioctl request, so that pending packets in the netmap ring are transferred to the transmission ring, if in native mode, or to the installed qdisc, if in generic mode. Also, as described further below, the sync thread is exploited to perform flow control and notify the BQL library about the amount of bytes that have been transferred to the network device.

The read method is called by the reading thread to retrieve new incoming packets stored in the netmap receiver ring and pass them to the appropriate ns-3 protocol handler for further processing within the simulator’s network stack. After retrieving packets, the reading thread also synchronizes the netmap receiver ring, so that the retrieved packets can be removed from the netmap receiver ring.

The NetmapNetDevice also specializes the write method, i.e., the method used to transmit a packet received from the upper layer (the ns-3 traffic control layer). The write method uses the netmap API to write the packet to a free slot in the netmap transmission ring. After writing a packet, the write method checks whether there is enough room in the netmap transmission ring for another packet. If not, the NetmapNetDevice stops its queue so that the ns-3 traffic control layer does not attempt to send a packet that could not be stored in the netmap transmission ring.

A stopped NetmapNetDevice queue needs to be restarted as soon as some room is made in the netmap transmission ring. The sync thread can be exploited for this purpose, given that it periodically synchronizes the netmap transmission ring. In particular, the sync thread also checks the number of free slots in the netmap transmission ring in case the NetmapNetDevice queue is stopped. If the number of free slots exceeds a configurable value, the sync thread restarts the NetmapNetDevice queue and wakes the associated ns-3 qdisc. The NetmapNetDevice also supports BQL: the write method notifies the BQL library of the amount of bytes that have been written to the netmap transmission ring, while the sync thread notifies the BQL library of the amount of bytes that have been removed from the netmap transmission ring and transferred to the NIC since the previous notification.

Scope and Limitations¶

The main scope of NetmapNetDevice is to support the flow-control between the physical device and the upper layer and using at best the computational resources to process packets. However, the (Linux) system and network device must support netmap to make use of this feature.

Usage¶

The installation of netmap itself on a host machine is out of scope for this document. Refer to the netmap GitHub README for instructions.

The ns-3 netmap code has only been tested on Linux; it is not clear whether other operating systems can be supported.

If ns-3 is able to detect the presence of netmap on the system, it will report that:

Netmap emulation FdNetDevice  : not enabled

If not, it will report:

Netmap emulation FdNetDevice  : not enabled (needs net/netmap_user.h)

To run FdNetDevice-enabled simulations, one must pass the --enable-sudo option to ./waf configure, or else run the simulations with root privileges.

Helpers¶

ns-3 netmap support uses a NetMapNetDeviceHelper helper object to install the NetmapNetDevice. In other respects, the API and use is similar to that of the EmuFdNetDeviceHelper.

Attributes¶

There is one attribute specialized to NetmapNetDevice, named SyncAndNotifyQueuePeriod. This value takes an integer number of microseconds, and is used as the period of time after which the device syncs the netmap ring and notifies queue status. The value should be close to the interrupt coalescence period of the real device. Users may want to tune this parameter for their own system; it should be a compromise between CPU usage and accuracy in the ring sync (if it is too high, the device goes into starvation and lower throughput occurs).

Output¶

The NetmapNetDevice does not provide any specialized output, but supports the FdNetDevice output and traces (such as a promiscuous sniffer trace).

Examples¶

Several examples are provided:

Note that all the examples run in emulation mode through netmap (with NetmapNetDevice) and raw socket (with FdNetDevice).

DPDK NetDevice¶

Data Plane Development Kit (DPDK) is a library hosted by The Linux Foundation to accelerate packet processing workloads (https://www.dpdk.org/).

The DpdkNetDevice class provides the implementation of a network device which uses DPDK’s fast packet processing abilities and bypasses the kernel. This class is included in the src/fd-net-device model. The DpdkNetDevice class inherits the FdNetDevice class and overrides the functions which are required by ns-3 to interact with DPDK environment.

The DpdkNetDevice for ns-3 [Patel2019] was developed by Harsh Patel, Hrishikesh Hiraskar and Mohit P. Tahiliani. They were supported by Intel Technology India Pvt. Ltd., Bangalore for this work.

[Patel2019]

Harsh Patel, Hrishikesh Hiraskar, Mohit P. Tahiliani, “Extending Network Emulation Support in ns-3 using DPDK”, Proceedings of the 2019 Workshop on ns-3, ACM, Pages 17-24, (https://dl.acm.org/doi/abs/10.1145/3321349.3321358)

Model Description¶

DpdkNetDevice is a network device which provides network emulation capabilities i.e. to allow simulated nodes to interact with real hosts and vice versa. The main feature of the DpdkNetDevice is that is uses the Environment Abstraction Layer (EAL) provided by DPDK to perform fast packet processing. EAL hides the device specific attributes from the applications and provides an interface via which the applications can interact directly with the Network Interface Card (NIC). This allows ns-3 to send/receive packets directly to/from the NIC without the kernel involvement.

Design¶

DpdkNetDevice is designed to act as an interface between ns-3 and DPDK environment. There are 3 main phases in the life cycle of DpdkNetDevice:

Initialization¶

DpdkNetDeviceHelper model is responsible for the initialization of DpdkNetDevice. After this, the EAL is initialized, a memory pool is allocated, access to the Ethernet port is obtained and it is initialized, reception (Rx) and transmission (Tx) queues are set up on the port, Rx and Tx buffers are set up and LaunchCore method is called which will launch the HandleRx method to handle reading of packets in burst.

Packet Transfer¶

DPDK interacts with packet in the form of mbuf, a data structure provided by it, while ns-3 interacts with packets in the form of raw buffer. The packet transfer functions take care of converting DPDK mbufs to ns-3 buffers. The functions are read and write.

Termination¶

When ns-3 is done using DpdkNetDevice, the DpdkNetDevice will stop polling for Rx, free the allocated mbuf packets and then the mbuf pool. Lastly, it will stop the Ethernet device and close the port.

Scope and Limitations¶

The current implementation supports only one NIC to be bound to DPDK with single Rx and Tx on the NIC. This can be extended to support multiple NICs and multiple Rx/Tx queues simultaneously. Currently there is no support for Jumbo frames, which can be added. Offloading, scheduling features can also be added. Flow control and support for qdisc can be added to provide a more extensive model for network testing.

DPDK Installation¶

This section contains information on downloading DPDK source code and setting up DPDK for DpdkNetDevice to work.

Is my NIC supported by DPDK?¶

Check Supported Devices.

Not supported? Use Virtual Machine instead¶

Install Oracle VM VirtualBox. Create a new VM and install Ubuntu on it. Open settings, create a network adapter with following configuration:

Then rest of the steps are same as follows.

DPDK can be installed in 2 ways:

Install DPDK on Ubuntu¶

To install DPDK on Ubuntu, run the following command:

apt-get install dpdk dpdk-dev libdpdk-dev dpdk-igb-uio-dkms

Ubuntu 20.04 has packaged DPDK v19.11 LTS which is tested with this module and DpdkNetDevice will only be enabled if this version is available.

Compile from Source¶

To compile DPDK from source, you need to perform the following 4 steps:

1. Download the source¶

Visit the DPDK Downloads page to download the latest stable source. (This module has been tested with version 19.11 LTS and DpdkNetDevice will only be enabled if this version is available.)

2. Configure DPDK as a shared library¶

In the DPDK directory, edit the config/common_base file to change the following line to compile DPDK as a shared library:

# Compile to share library
CONFIG_RTE_BUILD_SHARED_LIB=y

3. Install the source¶

Refer to Installation for detailed instructions.

For a 64 bit linux machine with gcc, run:

make install T=x86_64-native-linuxapp-gcc DESTDIR=install

4. Export DPDK Environment variables¶

Export the following environment variables:

For example:

export RTE_SDK=/home/username/dpdk/dpdk-stable-19.11.1
export RTE_TARGET=x86_64-native-linuxapp-gcc

(Note: In case DPDK is moved, ns-3 needs to be reconfigured using ./waf configure [options])

It is advisable that you export these variables in .bashrc or similar for reusability.

Load DPDK Drivers to kernel¶

Execute the following:

sudo modprobe uio_pci_generic
sudo modprobe uio
sudo modprobe vfio-pci
sudo modprobe igb_uio # for ubuntu package
# OR
sudo insmod $RTE_SDK/$RTE_TARGET/kmod/igb_uio.ko # for dpdk source

These should be done every time you reboot your system.

Configure hugepages¶

Refer System Requirements for detailed instructions.

To allocate hugepages at runtime, write a value such as ‘256’ to the following:

echo 256 > /sys/kernel/mm/hugepages/hugepages-2048kB/nr_hugepages

To allocate hugepages at boot time, edit /etc/default/grub, and following to GRUB_CMDLINE_LINUX_DEFAULT:

hugepages=256

We suggest minimum of number of 256 to run our applications. (This is to test an application run at 1 Gbps on a 1 Gbps NIC.) You can use any number of hugepages based on your system capacity and application requirements.

Then update the grub configurations using:

sudo update-grub

OR

sudo update-grub2

You will need to reboot your system in order to see these changes.

To check allocation of hugepages, run:

cat /proc/meminfo | grep HugePages

You will see the number of hugepages allocated, they should be equal to the number you used above.

Once the hugepage memory is reserved (at either runtime or boot time), to make the memory available for DPDK use, perform the following steps:

sudo mkdir /mnt/huge
sudo mount -t hugetlbfs nodev /mnt/huge

The mount point can be made permanent across reboots, by adding the following line to the /etc/fstab file:

nodev /mnt/huge hugetlbfs defaults 0 0

Usage¶

The status of DPDK support is shown in the output of ./waf configure. If it is found, a user should see:

DPDK NetDevice                : enabled

DpdkNetDeviceHelper class supports the configuration of DpdkNetDevice.

+----------------------+
|         host 1       |
+----------------------+
|   ns-3 simulation    |
+----------------------+
|       ns-3 Node      |
|  +----------------+  |
|  |    ns-3 TCP    |  |
|  +----------------+  |
|  |    ns-3 IP     |  |
|  +----------------+  |
|  |  DpdkNetDevice |  |
|  |    10.1.1.1    |  |
|  +----------------+  |
|  |   raw socket   |  |
|--+----------------+--|
|       | eth0 |       |
+-------+------+-------+


        10.1.1.11


            |


            +-------------- ( Internet ) ----

Initialization of DPDK driver requires initialization of EAL. EAL requires PMD (Poll Mode Driver) Library for using NIC. DPDK supports multiple Poll Mode Drivers and you can use one that works for your NIC. PMD Library can be set via DpdkNetDeviceHelper::SetPmdLibrary, as follows:

DpdkNetDeviceHelper* dpdk = new DpdkNetDeviceHelper ();
dpdk->SetPmdLibrary("librte_pmd_e1000.so");

Also, NIC should be bound to DPDK Driver in order to be used with EAL. The default driver used is uio_pci_generic which supports most of the NICs. You can change it using DpdkNetDeviceHelper::SetDpdkDriver, as follows:

DpdkNetDeviceHelper* dpdk = new DpdkNetDeviceHelper ();
dpdk->SetDpdkDriver("igb_uio");

Attributes¶

The DpdkNetDevice provides a number of attributes:

Note: Default values work well with 1Gbps traffic.

Output¶

As DpdkNetDevice is inherited from FdNetDevice, all the output methods provided by FdNetDevice can be used directly.

Examples¶

The following examples are provided:

Tap NetDevice¶

The Tap NetDevice can be used to allow a host system or virtual machines to interact with a simulation.

TapBridge Model Overview¶

The Tap Bridge is designed to integrate “real” internet hosts (or more precisely, hosts that support Tun/Tap devices) into ns-3 simulations. The goal is to make it appear to a “real” host node in that it has an ns-3 net device as a local device. The concept of a “real host” is a bit slippery since the “real host” may actually be virtualized using readily available technologies such as VMware, VirtualBox or OpenVZ.

Since we are, in essence, connecting the inputs and outputs of an ns-3 net device to the inputs and outputs of a Linux Tap net device, we call this arrangement a Tap Bridge.

There are three basic operating modes of this device available to users. Basic functionality is essentially identical, but the modes are different in details regarding how the arrangement is created and configured; and what devices can live on which side of the bridge.

We call these three modes the ConfigureLocal, UseLocal and UseBridge modes. The first “word” in the camel case mode identifier indicates who has the responsibility for creating and configuring the taps. For example, the “Configure” in ConfigureLocal mode indicates that it is the TapBridge that has responsibility for configuring the tap. In UseLocal mode and UseBridge modes, the “Use” prefix indicates that the TapBridge is asked to “Use” an existing configuration.

In other words, in ConfigureLocal mode, the TapBridge has the responsibility for creating and configuring the TAP devices. In UseBridge or UseLocal modes, the user provides a configuration and the TapBridge adapts to that configuration.

TapBridge ConfigureLocal Mode¶

In the ConfigureLocal mode, the configuration of the tap device is ns-3 configuration-centric. Configuration information is taken from a device in the ns-3 simulation and a tap device matching the ns-3 attributes is automatically created. In this case, a Linux computer is made to appear as if it was directly connected to a simulated ns-3 network.

This is illustrated below:

+--------+
|  Linux |
|  host  |                    +----------+
| ------ |                    |   ghost  |
|  apps  |                    |   node   |
| ------ |                    | -------- |
|  stack |                    |    IP    |     +----------+
| ------ |                    |   stack  |     |   node   |
|  TAP   |                    |==========|     | -------- |
| device | <----- IPC ------> |   tap    |     |    IP    |
+--------+                    |  bridge  |     |   stack  |


                              | -------- |     | -------- |


                              |   ns-3   |     |   ns-3   |


                              |   net    |     |   net    |


                              |  device  |     |  device  |


                              +----------+     +----------+


                                   ||               ||


                              +---------------------------+


                              |        ns-3 channel       |


                              +---------------------------+

In this case, the “ns-3 net device” in the “ghost node” appears as if it were actually replacing the TAP device in the Linux host. The ns-3 simulation creates the TAP device on the underlying Linux OS and configures the IP and MAC addresses of the TAP device to match the values assigned to the simulated ns-3 net device. The “IPC” link shown above is the network tap mechanism in the underlying OS. The whole arrangement acts as a conventional bridge; but a bridge between devices that happen to have the same shared MAC and IP addresses.

Here, the user is not required to provide any configuration information specific to the tap. A tap device will be created and configured by ns-3 according to its defaults, and the tap device will have its name assigned by the underlying operating system according to its defaults.

If the user has a requirement to access the created tap device, he or she may optionally provide a “DeviceName” attribute. In this case, the created OS tap device will be named accordingly.

The ConfigureLocal mode is the default operating mode of the Tap Bridge.

TapBridge UseLocal Mode¶

The UseLocal mode is quite similar to the ConfigureLocal mode. The significant difference is, as the mode name implies, the TapBridge is going to “Use” an existing tap device previously created and configured by the user. This mode is particularly useful when a virtualization scheme automatically creates tap devices and ns-3 is used to provide simulated networks for those devices.

+--------+
|  Linux |
|  host  |                    +----------+
| ------ |                    |   ghost  |
|  apps  |                    |   node   |
| ------ |                    | -------- |
|  stack |                    |    IP    |     +----------+
| ------ |                    |   stack  |     |   node   |
|  TAP   |                    |==========|     | -------- |
| device | <----- IPC ------> |   tap    |     |    IP    |
| MAC X  |                    |  bridge  |     |   stack  |
+--------+                    | -------- |     | -------- |


                              |   ns-3   |     |   ns-3   |


                              |   net    |     |   net    |


                              |  device  |     |  device  |


                              |  MAC Y   |     |  MAC Z   |


                              +----------+     +----------+


                                   ||               ||


                              +---------------------------+


                              |        ns-3 channel       |


                              +---------------------------+

In this case, the pre-configured MAC address of the “Tap device” (MAC X) will not be the same as that of the bridged “ns-3 net device” (MAC Y) shown in the illustration above. In order to bridge to ns-3 net devices which do not support SendFrom() (especially wireless STA nodes) we impose a requirement that only one Linux device (with one unique MAC address – here X) generates traffic that flows across the IPC link. This is because the MAC addresses of traffic across the IPC link will be “spoofed” or changed to make it appear to Linux and ns-3 that they have the same address. That is, traffic moving from the Linux host to the ns-3 ghost node will have its MAC address changed from X to Y and traffic from the ghost node to the Linux host will have its MAC address changed from Y to X. Since there is a one-to-one correspondence between devices, there may only be one MAC source flowing from the Linux side. This means that Linux bridges with more than one net device added are incompatible with UseLocal mode.

In UseLocal mode, the user is expected to create and configure a tap device completely outside the scope of the ns-3 simulation using something like:

$ sudo tunctl -t tap0
$ sudo ifconfig tap0 hw ether 08:00:2e:00:00:01
$ sudo ifconfig tap0 10.1.1.1 netmask 255.255.255.0 up

To tell the TapBridge what is going on, the user will set either directly into the TapBridge or via the TapBridgeHelper, the “DeviceName” attribute. In the case of the configuration above, the “DeviceName” attribute would be set to “tap0” and the “Mode” attribute would be set to “UseLocal”.

One particular use case for this mode is in the OpenVZ environment. There it is possible to create a Tap device on the “Hardware Node” and move it into a Virtual Private Server. If the TapBridge is able to use an existing tap device it is then possible to avoid the overhead of an OS bridge in that environment.

TapBridge UseBridge Mode¶

The simplest mode for those familiar with Linux networking is the UseBridge mode. Again, the “Use” prefix indicates that the TapBridge is going to Use an existing configuration. In this case, the TapBridge is going to logically extend a Linux bridge into ns-3.

This is illustrated below:

+---------+
|  Linux  |                             +----------+
| ------- |                             |   ghost  |
|  apps   |                             |   node   |
| ------- |                             | -------- |
|  stack  |                             |    IP    |     +----------+
| ------- | +--------+                  |   stack  |     |   node   |
| Virtual | |  TAP   |                  |==========|     | -------- |
| Device  | | Device | <---- IPC -----> |   tap    |     |    IP    |
+---------+ +--------+                  |  bridge  |     |   stack  |


    ||          ||                      | -------- |     | -------- |
+--------------------+                  |   ns-3   |     |   ns-3   |
| OS (brctl) Bridge  |                  |   net    |     |   net    |
+--------------------+                  |  device  |     |  device  |


                                        +----------+     +----------+


                                             ||               ||


                                        +---------------------------+


                                        |        ns-3 channel       |


                                        +---------------------------+

In this case, a computer running Linux applications, protocols, etc., is connected to a ns-3 simulated network in such a way as to make it appear to the Linux host that the TAP device is a real network device participating in the Linux bridge.

In the ns-3 simulation, a TapBridge is created to match each TAP Device. The name of the TAP Device is assigned to the Tap Bridge using the “DeviceName” attribute. The TapBridge then logically extends the OS bridge to encompass the ns-3 net device.

Since this mode logically extends an OS bridge, there may be many Linux net devices on the non-ns-3 side of the bridge. Therefore, like a net device on any bridge, the ns-3 net device must deal with the possibly of many source addresses. Thus, ns-3 devices must support SendFrom() (NetDevice::SupportsSendFrom() must return true) in order to be configured for use in UseBridge mode.

It is expected that the user will do something like the following to configure the bridge and tap completely outside ns-3:

$ sudo brctl addbr mybridge
$ sudo tunctl -t mytap
$ sudo ifconfig mytap hw ether 00:00:00:00:00:01
$ sudo ifconfig mytap 0.0.0.0 up
$ sudo brctl addif mybridge mytap
$ sudo brctl addif mybridge ...
$ sudo ifconfig mybridge 10.1.1.1 netmask 255.255.255.0 up

To tell the TapBridge what is going on, the user will set either directly into the TapBridge or via the TapBridgeHelper, the “DeviceName” attribute. In the case of the configuration above, the “DeviceName” attribute would be set to “mytap” and the “Mode” attribute would be set to “UseBridge”.

This mode is especially useful in the case of virtualization where the configuration of the virtual hosts may be dictated by another system and not be changeable to suit ns-3. For example, a particular VM scheme may create virtual “vethx” or “vmnetx” devices that appear local to virtual hosts. In order to connect to such systems, one would need to manually create TAP devices on the host system and brigde these TAP devices to the existing (VM) virtual devices. The job of the Tap Bridge in this case is to extend the bridge to join a ns-3 net device.

TapBridge ConfigureLocal Operation¶

In ConfigureLocal mode, the TapBridge and therefore its associated ns-3 net device appears to the Linux host computer as a network device just like any arbitrary “eth0” or “ath0” might appear. The creation and configuration of the TAP device is done by the ns-3 simulation and no manual configuration is required by the user. The IP addresses, MAC addresses, gateways, etc., for created TAP devices are extracted from the simulation itself by querying the configuration of the ns-3 device and the TapBridge Attributes.

Since the MAC addresses are identical on the Linux side and the ns-3 side, we can use Send() on the ns-3 device which is available on all ns-3 net devices. Since the MAC addresses are identical there is no requirement to hook the promiscuous callback on the receive side. Therefore there are no restrictions on the kinds of net device that are usable in ConfigureLocal mode.

The TapBridge appears to an ns-3 simulation as a channel-less net device. This device must not have an IP address associated with it, but the bridged (ns-3) net device must have an IP address. Be aware that this is the inverse of an ns-3 BridgeNetDevice (or a conventional bridge in general) which demands that its bridge ports not have IP addresses, but allows the bridge device itself to have an IP address.

The host computer will appear in a simulation as a “ghost” node that contains one TapBridge for each NetDevice that is being bridged. From the perspective of a simulation, the only difference between a ghost node and any other node will be the presence of the TapBridge devices. Note however, that the presence of the TapBridge does affect the connectivity of the net device to the IP stack of the ghost node.

Configuration of address information and the ns-3 devices is not changed in any way if a TapBridge is present. A TapBridge will pick up the addressing information from the ns-3 net device to which it is connected (its “bridged” net device) and use that information to create and configure the TAP device on the real host.

The end result of this is a situation where one can, for example, use the standard ping utility on a real host to ping a simulated ns-3 node. If correct routes are added to the internet host (this is expected to be done automatically in future ns-3 releases), the routing systems in ns-3 will enable correct routing of the packets across simulated ns-3 networks. For an example of this, see the example program, tap-wifi-dumbbell.cc in the ns-3 distribution.

The Tap Bridge lives in a kind of a gray world somewhere between a Linux host and an ns-3 bridge device. From the Linux perspective, this code appears as the user mode handler for a TAP net device. In ConfigureLocal mode, this Tap device is automatically created by the ns-3 simulation. When the Linux host writes to one of these automatically created /dev/tap devices, the write is redirected into the TapBridge that lives in the ns-3 world; and from this perspective, the packet write on Linux becomes a packet read in the Tap Bridge. In other words, a Linux process writes a packet to a tap device and this packet is redirected by the network tap mechanism toan ns-3 process where it is received by the TapBridge as a result of a read operation there. The TapBridge then writes the packet to the ns-3 net device to which it is bridged; and therefore it appears as if the Linux host sent a packet directly through an ns-3 net device onto an ns-3 network.

In the other direction, a packet received by the ns-3 net device connected to the Tap Bridge is sent via a receive callback to the TapBridge. The TapBridge then takes that packet and writes it back to the host using the network tap mechanism. This write to the device will appear to the Linux host as if a packet has arrived on a net device; and therefore as if a packet received by the ns-3 net device during a simulation has appeared on a real Linux net device.

The upshot is that the Tap Bridge appears to bridge a tap device on a Linux host in the “real world” to an ns-3 net device in the simulation. Because the TAP device and the bridged ns-3 net device have the same MAC address and the network tap IPC link is not externalized, this particular kind of bridge makes it appear that a ns-3 net device is actually installed in the Linux host.

In order to implement this on the ns-3 side, we need a “ghost node” in the simulation to hold the bridged ns-3 net device and the TapBridge. This node should not actually do anything else in the simulation since its job is simply to make the net device appear in Linux. This is not just arbitrary policy, it is because:

Of course, if you understand all of the issues you can take control of your own destiny and do whatever you want – we do not actively prevent you from using the ghost node for anything you decide. You will be able to perform typical ns-3 operations on the ghost node if you so desire. The internet stack, for example, must be there and functional on that node in order to participate in IP address assignment and global routing. However, as mentioned above, interfaces talking to any TapBridge or associated bridged net devices will not work completely. If you understand exactly what you are doing, you can set up other interfaces and devices on the ghost node and use them; or take advantage of the operational send side of the bridged devices to create traffic generators. We generally recommend that you treat this node as a ghost of the Linux host and leave it to itself, though.

TapBridge UseLocal Mode Operation¶

As described in above, the TapBridge acts like a bridge from the “real” world into the simulated ns-3 world. In the case of the ConfigureLocal mode, life is easy since the IP address of the Tap device matches the IP address of the ns-3 device and the MAC address of the Tap device matches the MAC address of the ns-3 device; and there is a one-to-one relationship between the devices.

Things are slightly complicated when a Tap device is externally configured with a different MAC address than the ns-3 net device. The conventional way to deal with this kind of difference is to use promiscuous mode in the bridged device to receive packets destined for the different MAC address and forward them off to Linux. In order to move packets the other way, the conventional solution is SendFrom() which allows a caller to “spoof” or change the source MAC address to match the different Linux MAC address.

We do have a specific requirement to be able to bridge Linux Virtual Machines onto wireless STA nodes. Unfortunately, the 802.11 spec doesn’t provide a good way to implement SendFrom(), so we have to work around that problem.

To this end, we provided the UseLocal mode of the Tap Bridge. This mode allows you approach the problem as if you were creating a bridge with a single net device. A single allowed address on the Linux side is remembered in the TapBridge, and all packets coming from the Linux side are repeated out the ns-3 side using the ns-3 device MAC source address. All packets coming in from the ns-3 side are repeated out the Linux side using the remembered MAC address. This allows us to use Send() on the ns-3 device side which is available on all ns-3 net devices.

UseLocal mode is identical to the ConfigureLocal mode except for the creation and configuration of the tap device and the MAC address spoofing.

TapBridge UseBridge Operation¶

As described in the ConfigureLocal mode section, when the Linux host writes to one of the /dev/tap devices, the write is redirected into the TapBridge that lives in the ns-3 world. In the case of the UseBridge mode, these packets will need to be sent out on the ns-3 network as if they were sent on a device participating in the Linux bridge. This means calling the SendFrom() method on the bridged device and providing the source MAC address found in the packet.

In the other direction, a packet received by an ns-3 net device is hooked via callback to the TapBridge. This must be done in promiscuous mode since the goal is to bridge the ns-3 net device onto the OS (brctl) bridge of which the TAP device is a part.

For these reasons, only ns-3 net devices that support SendFrom() and have a hookable promiscuous receive callback are allowed to participate in UseBridge mode TapBridge configurations.

Tap Bridge Channel Model¶

There is no channel model associated with the Tap Bridge. In fact, the intention is make it appear that the real internet host is connected to the channel of the bridged net device.

Tap Bridge Tracing Model¶

Unlike most ns-3 devices, the TapBridge does not provide any standard trace sources. This is because the bridge is an intermediary that is essentially one function call away from the bridged device. We expect that the trace hooks in the bridged device will be sufficient for most users,

Using the TapBridge¶

We expect that most users will interact with the TapBridge device through the TapBridgeHelper. Users of other helper classes, such as CSMA or Wifi, should be comfortable with the idioms used there.

Model Description¶

The source code for the Energy Framework is currently at: src/energy.

Design¶

The ns-3 Energy Framework is composed of 3 parts: Energy Source, Device Energy Model and Energy Harvester. The framework is implemented into the src/energy/models folder.

Energy Source¶

The Energy Source represents the power supply on each node. A node can have one or more energy sources, and each energy source can be connected to multiple device energy models. Connecting an energy source to a device energy model implies that the corresponding device draws power from the source. The basic functionality of the Energy Source is to provide energy for devices on the node. When energy is completely drained from the Energy Source, it notifies the devices on node such that each device can react to this event. Further, each node can access the Energy Source Objects for information such as remaining energy or energy fraction (battery level). This enables the implementation of energy aware protocols in ns-3.

In order to model a wide range of power supplies such as batteries, the Energy Source must be able to capture characteristics of these supplies. There are 2 important characteristics or effects related to practical batteries:

In order to incorporate the Rate Capacity Effect, the Energy Source uses current draw from all the devices on the same node to calculate energy consumption. Moreover, multiple Energy Harvesters can be connected to the Energy Source in order to replenish its energy. The Energy Source periodically polls all the devices and energy harvesters on the same node to calculate the total current drain and hence the energy consumption. When a device changes state, its corresponding Device Energy Model will notify the Energy Source of this change and new total current draw will be calculated. Similarly, every Energy Harvester update triggers an update to the connected Energy Source.

The Energy Source base class keeps a list of devices (Device Energy Model objects) and energy harvesters (Energy Harvester objects) that are using the particular Energy Source as power supply. When energy is completely drained, the Energy Source will notify all devices on this list. Each device can then handle this event independently, based on the desired behavior that should be followed in case of power outage.

Device Energy Model¶

The Device Energy Model is the energy consumption model of a device installed on the node. It is designed to be a state based model where each device is assumed to have a number of states, and each state is associated with a power consumption value. Whenever the state of the device changes, the corresponding Device Energy Model will notify the Energy Source of the new current draw of the device. The Energy Source will then calculate the new total current draw and update the remaining energy.

The Device Energy Model can also be used for devices that do not have finite number of states. For example, in an electric vehicle, the current draw of the motor is determined by its speed. Since the vehicle’s speed can take continuous values within a certain range, it is infeasible to define a set of discrete states of operation. However, by converting the speed value into current directly, the same set of Device Energy Model APIs can still be used.

Energy Harvester¶

The energy harvester represents the elements that harvest energy from the environment and recharge the Energy Source to which it is connected. The energy harvester includes the complete implementation of the actual energy harvesting device (e.g., a solar panel) and the environment (e.g., the solar radiation). This means that in implementing an energy harvester, the energy contribution of the environment and the additional energy requirements of the energy harvesting device such as the conversion efficiency and the internal power consumption of the device needs to be jointly modeled.

WiFi Radio Energy Model¶

The WiFi Radio Energy Model is the energy consumption model of a Wifi net device. It provides a state for each of the available states of the PHY layer: Idle, CcaBusy, Tx, Rx, ChannelSwitch, Sleep, Off. Each of such states is associated with a value (in Ampere) of the current draw (see below for the corresponding attribute names). A Wifi Radio Energy Model PHY Listener is registered to the Wifi PHY in order to be notified of every Wifi PHY state transition. At every transition, the energy consumed in the previous state is computed and the energy source is notified in order to update its remaining energy.

The Wifi Tx Current Model gives the possibility to compute the current draw in the transmit state as a function of the nominal tx power (in dBm), as observed in several experimental measurements. To this purpose, the Wifi Radio Energy Model PHY Listener is notified of the nominal tx power used to transmit the current frame and passes such a value to the Wifi Tx Current Model which takes care of updating the current draw in the Tx state. Hence, the energy consumption is correctly computed even if the Wifi Remote Station Manager performs per-frame power control. Currently, a Linear Wifi Tx Current Model is implemented which computes the tx current as a linear function (according to parameters that can be specified by the user) of the nominal tx power in dBm.

The Wifi Radio Energy Model offers the possibility to specify a callback that is invoked when the energy source is depleted. If such a callback is not specified when the Wifi Radio Energy Model Helper is used to install the model on a device, a callback is implicitly made so that the Wifi PHY is put in the OFF mode (hence no frame is transmitted nor received afterwards) when the energy source is depleted. Likewise, it is possible to specify a callback that is invoked when the energy source is recharged (which might occur in case an energy harvester is connected to the energy source). If such a callback is not specified when the Wifi Radio Energy Model Helper is used to install the model on a device, a callback is implicitly made so that the Wifi PHY is resumed from the OFF mode when the energy source is recharged.

Future Work¶

For Device Energy Models, we are planning to include support for other PHY layer models provided in ns-3 such as WiMAX, and to model the energy consumptions of other non communicating devices, like a generic sensor and a CPU. For Energy Sources, we are planning to included new types of Energy Sources such as Supercapacitor and Nickel-Metal Hydride (Ni-MH) battery. For the Energy Harvesters, we are planning to implement an energy harvester that recharges the energy sources according to the power levels defined in a user customizable dataset of real measurements.

References¶

[1]

ns-2 Energy model: http://www.cubinlab.ee.unimelb.edu.au/~jrid/Docs/Manuel-NS2/node204.html

[2]

H. Wu, S. Nabar and R. Poovendran. An Energy Framework for the Network Simulator 3 (ns-3).

[3]

M. Handy and D. Timmermann. Simulation of mobile wireless networks with accurate modelling of non-linear battery effects. In Proc. of Applied simulation and Modeling (ASM), 2003.

[4]

D. N. Rakhmatov and S. B. Vrudhula. An analytical high-level battery model for use in energy management of portable electronic systems. In Proc. of IEEE/ACM International Conference on Computer Aided Design (ICCAD’01), pages 488-493, November 2001.

[5]

D. N. Rakhmatov, S. B. Vrudhula, and D. A. Wallach. Battery lifetime prediction for energy-aware computing. In Proc. of the 2002 International Symposium on Low Power Electronics and Design (ISLPED’02), pages 154-159, 2002.

[6]

C. Tapparello, H. Ayatollahi and W. Heinzelman. Extending the Energy Framework for Network Simulator 3 (ns-3). Workshop on ns-3 (WNS3), Poster Session, Atlanta, GA, USA. May, 2014.

[7]

C. Tapparello, H. Ayatollahi and W. Heinzelman. Energy Harvesting Framework for Network Simulator 3 (ns-3). 2nd International Workshop on Energy Neutral Sensing Systems (ENSsys), Memphis, TN, USA. November 6, 2014.

Usage¶

The main way that ns-3 users will typically interact with the Energy Framework is through the helper API and through the publicly visible attributes of the framework. The helper API is defined in src/energy/helper/*.h.

In order to use the energy framework, the user must install an Energy Source for the node of interest, the corresponding Device Energy Model for the network devices and, if necessary, the one or more Energy Harvester. Energy Source (objects) are aggregated onto each node by the Energy Source Helper. In order to allow multiple energy sources per node, we aggregate an Energy Source Container rather than directly aggregating a source object.

The Energy Source object keeps a list of Device Energy Model and Energy Harvester objects using the source as power supply. Device Energy Model objects are installed onto the Energy Source by the Device Energy Model Helper, while Energy Harvester object are installed by the Energy Harvester Helper. User can access the Device Energy Model objects through the Energy Source object to obtain energy consumption information of individual devices. Moreover, the user can access to the Energy Harvester objects in order to gather information regarding the current harvestable power and the total energy harvested by the harvester.

Examples¶

The example directories, src/examples/energy and examples/energy, contain some basic code that shows how to set up the framework.

Helpers¶

Energy Source Helper¶

Base helper class for Energy Source objects, this helper Aggregates Energy Source object onto a node. Child implementation of this class creates the actual Energy Source object.

Device Energy Model Helper¶

Base helper class for Device Energy Model objects, this helper attaches Device Energy Model objects onto Energy Source objects. Child implementation of this class creates the actual Device Energy Model object.

Energy Harvesting Helper¶

Base helper class for Energy Harvester objects, this helper attaches Energy Harvester objects onto Energy Source objects. Child implementation of this class creates the actual Energy Harvester object.

Attributes¶

Attributes differ between Energy Sources, Devices Energy Models and Energy Harvesters implementations, please look at the specific child class for details.

Basic Energy Source¶

RV Battery Model¶

WiFi Radio Energy Model¶

Basic Energy Harvester¶

Tracing¶

Traced values differ between Energy Sources, Devices Energy Models and Energy Harvesters implementations, please look at the specific child class for details.

Basic Energy Source¶

RV Battery Model¶

WiFi Radio Energy Model¶

Basic Energy Harvester¶

Validation¶

Comparison of the Energy Framework against actual devices have not been performed. Current implementation of the Energy Framework is checked numerically for computation errors. The RV battery model is validated by comparing results with what was presented in the original RV battery model paper.

Model Description¶

The source code for the new module lives in the directory src/flow-monitor.

The Flow Monitor module goal is to provide a flexible system to measure the performance of network protocols. The module uses probes, installed in network nodes, to track the packets exchanged by the nodes, and it will measure a number of parameters. Packets are divided according to the flow they belong to, where each flow is defined according to the probe’s characteristics (e.g., for IP, a flow is defined as the packets with the same {protocol, source (IP, port), destination (IP, port)} tuple.

The statistics are collected for each flow can be exported in XML format. Moreover, the user can access the probes directly to request specific stats about each flow.

Design¶

Flow Monitor module is designed in a modular way. It can be extended by subclassing ns3::FlowProbe and ns3::FlowClassifier. Typically, a subclass of ns3::FlowProbe works by listening to the appropriate class Traces, and then uses its own ns3::FlowClassifier subclass to classify the packets passing though each node.

Each Probe can try to listen to other classes traces (e.g., ns3::Ipv4FlowProbe will try to use any ns3::NetDevice trace named TxQueue/Drop) but this is something that the user should not rely into blindly, because the trace is not guaranteed to be in every type of ns3::NetDevice. As an example, CsmaNetDevice and PointToPointNetDevice have a TxQueue/Drop trace, while WiFiNetDevice does not.

The full module design is described in [FlowMonitor]

Scope and Limitations¶

At the moment, probes and classifiers are available only for IPv4 and IPv6.

IPv4 and IPv6 probes will classify packets in four points:

Since the packets are tracked at IP level, any retransmission caused by L4 protocols (e.g., TCP) will be seen by the probe as a new packet.

A Tag will be added to the packet (ns3::Ipv[4,6]FlowProbeTag). The tag will carry basic packet’s data, useful for the packet’s classification.

It must be underlined that only L4 (TCP, UDP) packets are, so far, classified. Moreover, only unicast packets will be classified. These limitations may be removed in the future.

The data collected for each flow are:

It is worth pointing out that the probes measure the packet bytes including IP headers. The L2 headers are not included in the measure.

These stats will be written in XML form upon request (see the Usage section).

The “lost” packets problem¶

At the end of a simulation, Flow Monitor could report about “lost” packets, i.e., packets that Flow Monitor have lost track of.

It is important to keep in mind that Flow Monitor records the packets statistics by intercepting them at a given network level - let’s say at IP level. When the simulation ends, any packet queued for transmission below the IP level will be considered as lost.

It is strongly suggested to consider this point when using Flow Monitor. The user can choose to:

The second method is the suggested one. Usually a few seconds are enough (the exact value depends on the network type).

It is important to stress that “lost” packets could be anywhere in the network, and could count toward the received packets or the dropped ones. Ideally, their number should be zero or a minimal fraction of the other ones, i.e., they should be “statistically irrelevant”.

References¶

[FlowMonitor]

Usage¶

The module usage is extremely simple. The helper will take care of about everything.

The typical use is:

// Flow monitor
Ptr<FlowMonitor> flowMonitor;
FlowMonitorHelper flowHelper;
flowMonitor = flowHelper.InstallAll();
-yourApplicationsContainer-.Stop (Seconds (stop_time));;
Simulator::Stop (Seconds(stop_time+cleanup_time));
Simulator::Run ();
flowMonitor->SerializeToXmlFile("NameOfFile.xml", true, true);

the SerializeToXmlFile () function 2nd and 3rd parameters are used respectively to activate/deactivate the histograms and the per-probe detailed stats. Other possible alternatives can be found in the Doxygen documentation, while cleanup_time is the time needed by in-flight packets to reach their destinations.

Helpers¶

The helper API follows the pattern usage of normal helpers. Through the helper you can install the monitor in the nodes, set the monitor attributes, and print the statistics.

One important thing is: the ns3::FlowMonitorHelper must be instantiated only once in the main.

Attributes¶

The module provides the following attributes in ns3::FlowMonitor:

Output¶

The main model output is an XML formatted report about flow statistics. An example is:

<?xml version="1.0" ?>
<FlowMonitor>


  <FlowStats>


  <Flow flowId="1" timeFirstTxPacket="+0.0ns" timeFirstRxPacket="+20067198.0ns" timeLastTxPacket="+2235764408.0ns" timeLastRxPacket="+2255831606.0ns" delaySum="+138731526300.0ns" jitterSum="+1849692150.0ns" lastDelay="+20067198.0ns" txBytes="2149400" rxBytes="2149400" txPackets="3735" rxPackets="3735" lostPackets="0" timesForwarded="7466">


  </Flow>


  </FlowStats>


  <Ipv4FlowClassifier>


  <Flow flowId="1" sourceAddress="10.1.3.1" destinationAddress="10.1.2.2" protocol="6" sourcePort="49153" destinationPort="50000" />


  </Ipv4FlowClassifier>


  <Ipv6FlowClassifier>


  </Ipv6FlowClassifier>


  <FlowProbes>


  <FlowProbe index="0">


    <FlowStats  flowId="1" packets="3735" bytes="2149400" delayFromFirstProbeSum="+0.0ns" >


    </FlowStats>


  </FlowProbe>


  <FlowProbe index="2">


    <FlowStats  flowId="1" packets="7466" bytes="2224020" delayFromFirstProbeSum="+199415389258.0ns" >


    </FlowStats>


  </FlowProbe>


  <FlowProbe index="4">


    <FlowStats  flowId="1" packets="3735" bytes="2149400" delayFromFirstProbeSum="+138731526300.0ns" >


    </FlowStats>


  </FlowProbe>


  </FlowProbes>
</FlowMonitor>

The output was generated by a TCP flow from 10.1.3.1 to 10.1.2.2.

It is worth noticing that the index 2 probe is reporting more packets and more bytes than the other probes. That’s a perfectly normal behaviour, as packets are fragmented at IP level in that node.

It should also be observed that the receiving node’s probe (index 4) doesn’t count the fragments, as the reassembly is done before the probing point.

Examples¶

The examples are located in src/flow-monitor/examples.

Moreover, the following examples use the flow-monitor module:

Troubleshooting¶

Do not define more than one ns3::FlowMonitorHelper in the simulation.

Validation¶

The paper in the references contains a full description of the module validation against a test network.

Tests are provided to ensure the Histogram correct functionality.

Internet Stack¶

Internet stack aggregation¶

A bare class Node is not very useful as-is; other objects must be aggregated to it to provide useful node functionality.

The ns-3 source code directory src/internet provides implementation of TCP/IPv4- and IPv6-related components. These include IPv4, ARP, UDP, TCP, IPv6, Neighbor Discovery, and other related protocols.

Internet Nodes are not subclasses of class Node; they are simply Nodes that have had a bunch of IP-related objects aggregated to them. They can be put together by hand, or via a helper function InternetStackHelper::Install () which does the following to all nodes passed in as arguments:

void
InternetStackHelper::Install (Ptr<Node> node) const
{


  if (m_ipv4Enabled)


    {


      /* IPv4 stack */


      if (node->GetObject<Ipv4> () != 0)


        {


          NS_FATAL_ERROR ("InternetStackHelper::Install (): Aggregating "


                          "an InternetStack to a node with an existing Ipv4 object");


          return;


        }


      CreateAndAggregateObjectFromTypeId (node, "ns3::ArpL3Protocol");


      CreateAndAggregateObjectFromTypeId (node, "ns3::Ipv4L3Protocol");


      CreateAndAggregateObjectFromTypeId (node, "ns3::Icmpv4L4Protocol");


      // Set routing


      Ptr<Ipv4> ipv4 = node->GetObject<Ipv4> ();


      Ptr<Ipv4RoutingProtocol> ipv4Routing = m_routing->Create (node);


      ipv4->SetRoutingProtocol (ipv4Routing);


    }


  if (m_ipv6Enabled)


    {


      /* IPv6 stack */


      if (node->GetObject<Ipv6> () != 0)


        {


          NS_FATAL_ERROR ("InternetStackHelper::Install (): Aggregating "


                          "an InternetStack to a node with an existing Ipv6 object");


          return;


        }


      CreateAndAggregateObjectFromTypeId (node, "ns3::Ipv6L3Protocol");


      CreateAndAggregateObjectFromTypeId (node, "ns3::Icmpv6L4Protocol");


      // Set routing


      Ptr<Ipv6> ipv6 = node->GetObject<Ipv6> ();


      Ptr<Ipv6RoutingProtocol> ipv6Routing = m_routingv6->Create (node);


      ipv6->SetRoutingProtocol (ipv6Routing);


      /* register IPv6 extensions and options */


      ipv6->RegisterExtensions ();


      ipv6->RegisterOptions ();


    }


  if (m_ipv4Enabled || m_ipv6Enabled)


    {


      /* UDP and TCP stacks */


      CreateAndAggregateObjectFromTypeId (node, "ns3::UdpL4Protocol");


      node->AggregateObject (m_tcpFactory.Create<Object> ());


      Ptr<PacketSocketFactory> factory = CreateObject<PacketSocketFactory> ();


      node->AggregateObject (factory);


    }
}

Where multiple implementations exist in ns-3 (TCP, IP routing), these objects are added by a factory object (TCP) or by a routing helper (m_routing).

Note that the routing protocol is configured and set outside this function. By default, the following protocols are added:

void InternetStackHelper::Initialize ()
{


  SetTcp ("ns3::TcpL4Protocol");


  Ipv4StaticRoutingHelper staticRouting;


  Ipv4GlobalRoutingHelper globalRouting;


  Ipv4ListRoutingHelper listRouting;


  Ipv6ListRoutingHelper listRoutingv6;


  Ipv6StaticRoutingHelper staticRoutingv6;


  listRouting.Add (staticRouting, 0);


  listRouting.Add (globalRouting, -10);


  listRoutingv6.Add (staticRoutingv6, 0);


  SetRoutingHelper (listRouting);


  SetRoutingHelper (listRoutingv6);
}

By default, IPv4 and IPv6 are enabled.

Internet Node structure¶

An IP-capable Node (an ns-3 Node augmented by aggregation to have one or more IP stacks) has the following internal structure.

Layer-3 protocols¶

At the lowest layer, sitting above the NetDevices, are the “layer 3” protocols, including IPv4, IPv6, ARP and so on. The class Ipv4L3Protocol is an implementation class whose public interface is typically class Ipv4, but the Ipv4L3Protocol public API is also used internally at present.

In class Ipv4L3Protocol, one method described below is Receive ():

/**


  * Lower layer calls this method after calling L3Demux::Lookup


  * The ARP subclass needs to know from which NetDevice this


  * packet is coming to:


  *    - implement a per-NetDevice ARP cache


  *    - send back arp replies on the right device


  */
void Receive( Ptr<NetDevice> device, Ptr<const Packet> p, uint16_t protocol,
const Address &from, const Address &to, NetDevice::PacketType packetType);

First, note that the Receive () function has a matching signature to the ReceiveCallback in the class Node. This function pointer is inserted into the Node’s protocol handler when AddInterface () is called. The actual registration is done with a statement such as follows:

RegisterProtocolHandler ( MakeCallback (&Ipv4Protocol::Receive, ipv4),


                          Ipv4L3Protocol::PROT_NUMBER, 0);

The Ipv4L3Protocol object is aggregated to the Node; there is only one such Ipv4L3Protocol object. Higher-layer protocols that have a packet to send down to the Ipv4L3Protocol object can call GetObject<Ipv4L3Protocol> () to obtain a pointer, as follows:

Ptr<Ipv4L3Protocol> ipv4 = m_node->GetObject<Ipv4L3Protocol> ();
if (ipv4 != 0)


  {


    ipv4->Send (packet, saddr, daddr, PROT_NUMBER);


  }

This class nicely demonstrates two techniques we exploit in ns-3 to bind objects together: callbacks, and object aggregation.

Once IPv4 routing has determined that a packet is for the local node, it forwards it up the stack. This is done with the following function:

void
Ipv4L3Protocol::LocalDeliver (Ptr<const Packet> packet, Ipv4Header const&ip, uint32_t iif)

The first step is to find the right Ipv4L4Protocol object, based on IP protocol number. For instance, TCP is registered in the demux as protocol number 6. Finally, the Receive() function on the Ipv4L4Protocol (such as TcpL4Protocol::Receive is called.

We have not yet introduced the class Ipv4Interface. Basically, each NetDevice is paired with an IPv4 representation of such device. In Linux, this class Ipv4Interface roughly corresponds to the struct in_device; the main purpose is to provide address-family specific information (addresses) about an interface.

All the classes have appropriate traces in order to track sent, received and lost packets. The users is encouraged to use them so to find out if (and where) a packet is dropped. A common mistake is to forget the effects of local queues when sending packets, e.g., the ARP queue. This can be particularly puzzling when sending jumbo packets or packet bursts using UDP. The ARP cache pending queue is limited (3 datagrams) and IP packets might be fragmented, easily overfilling the ARP cache queue size. In those cases it is useful to increase the ARP cache pending size to a proper value, e.g.:

Config::SetDefault ("ns3::ArpCache::PendingQueueSize", UintegerValue (MAX_BURST_SIZE/L2MTU*3));

The IPv6 implementation follows a similar architecture. Dual-stacked nodes (one with support for both IPv4 and IPv6) will allow an IPv6 socket to receive IPv4 connections as a standard dual-stacked system does. A socket bound and listening to an IPv6 endpoint can receive an IPv4 connection and will return the remote address as an IPv4-mapped address. Support for the IPV6_V6ONLY socket option does not currently exist.

Layer-4 protocols and sockets¶

We next describe how the transport protocols, sockets, and applications tie together. In summary, each transport protocol implementation is a socket factory. An application that needs a new socket

For instance, to create a UDP socket, an application would use a code snippet such as the following:

Ptr<Udp> udpSocketFactory = GetNode ()->GetObject<Udp> ();
Ptr<Socket> m_socket = socketFactory->CreateSocket ();
m_socket->Bind (m_local_address);
...

The above will query the node to get a pointer to its UDP socket factory, will create one such socket, and will use the socket with an API similar to the C-based sockets API, such as Connect () and Send (). The address passed to the Bind (), Connect (), or Send () functions may be a Ipv4Address, Ipv6Address, or Address. If a Address is passed in and contains anything other than a Ipv4Address or Ipv6Address, these functions will return an error. The Bind (void) and Bind6 (void) functions bind to “0.0.0.0” and “::” respectively.

The socket can also be bound to a specific NetDevice though the BindToNetDevice (Ptr<NetDevice> netdevice) function. BindToNetDevice (Ptr<NetDevice> netdevice) will bind the socket to “0.0.0.0” and “::” (equivalent to calling Bind () and Bind6 (), unless the socket has been already bound to a specific address. Summarizing, the correct sequence is:

 Ptr<Udp> udpSocketFactory = GetNode ()->GetObject<Udp> ();


 Ptr<Socket> m_socket = socketFactory->CreateSocket ();


 m_socket->BindToNetDevice (n_netDevice);
...

or:

Ptr<Udp> udpSocketFactory = GetNode ()->GetObject<Udp> ();
Ptr<Socket> m_socket = socketFactory->CreateSocket ();
m_socket->Bind (m_local_address);
m_socket->BindToNetDevice (n_netDevice);
...

The following raises an error:

Ptr<Udp> udpSocketFactory = GetNode ()->GetObject<Udp> ();
Ptr<Socket> m_socket = socketFactory->CreateSocket ();
m_socket->BindToNetDevice (n_netDevice);
m_socket->Bind (m_local_address);
...

See the chapter on ns-3 sockets for more information.

We have described so far a socket factory (e.g. class Udp) and a socket, which may be specialized (e.g., class UdpSocket). There are a few more key objects that relate to the specialized task of demultiplexing a packet to one or more receiving sockets. The key object in this task is class Ipv4EndPointDemux. This demultiplexer stores objects of class Ipv4EndPoint. This class holds the addressing/port tuple (local port, local address, destination port, destination address) associated with the socket, and a receive callback. This receive callback has a receive function registered by the socket. The Lookup () function to Ipv4EndPointDemux returns a list of Ipv4EndPoint objects (there may be a list since more than one socket may match the packet). The layer-4 protocol copies the packet to each Ipv4EndPoint and calls its ForwardUp () method, which then calls the Receive () function registered by the socket.

An issue that arises when working with the sockets API on real systems is the need to manage the reading from a socket, using some type of I/O (e.g., blocking, non-blocking, asynchronous, …). ns-3 implements an asynchronous model for socket I/O; the application sets a callback to be notified of received data ready to be read, and the callback is invoked by the transport protocol when data is available. This callback is specified as follows:

void Socket::SetRecvCallback (Callback<void, Ptr<Socket>,


                              Ptr<Packet>,


                              const Address&> receivedData);

The data being received is conveyed in the Packet data buffer. An example usage is in class PacketSink:

m_socket->SetRecvCallback (MakeCallback(&PacketSink::HandleRead, this));

To summarize, internally, the UDP implementation is organized as follows:

IP-capable node interfaces¶

Many of the implementation details, or internal objects themselves, of IP-capable Node objects are not exposed at the simulator public API. This allows for different implementations; for instance, replacing the native ns-3 models with ported TCP/IP stack code.

The C++ public APIs of all of these objects is found in the src/network directory, including principally:

These are typically base class objects that implement the default values used in the implementation, implement access methods to get/set state variables, host attributes, and implement publicly-available methods exposed to clients such as CreateSocket.

Example path of a packet¶

These two figures show an example stack trace of how packets flow through the Internet Node objects.

IPv4¶

This chapter describes the ns-3 IPv4 address assignment and basic components tracking.

IPv4 addresses assignment¶

In order to use IPv4 on a network, the first thing to do is assigning IPv4 addresses.

Any IPv4-enabled ns-3 node will have at least one NetDevice: the ns3::LoopbackNetDevice. The loopback device address is 127.0.0.1. All the other NetDevices will have one (or more) IPv4 addresses.

Note that, as today, ns-3 does not have a NAT module, and it does not follows the rules about filtering private addresses (RFC 1918): 10.0.0.0/8, 172.16.0.0/12, and 192.168.0.0/16. These addresses are routed as any other address. This behaviour could change in the future.

IPv4 global addresses can be:

ns-3 can use both methods, and it’s quite important to understand the implications of both.

Manually assigned IPv4 addresses¶

This is probably the easiest and most used method. As an example:

Ptr<Node> n0 = CreateObject<Node> ();
Ptr<Node> n1 = CreateObject<Node> ();
NodeContainer net (n0, n1);
CsmaHelper csma;
NetDeviceContainer ndc = csma.Install (net);
NS_LOG_INFO ("Assign IPv4 Addresses.");
Ipv4AddressHelper ipv4;
ipv4.SetBase (Ipv4Address ("192.168.1.0"), NetMask ("/24"));
Ipv4InterfaceContainer ic = ipv4.Assign (ndc);

This method will add two global IPv4 addresses to the nodes.

Note that the addresses are assigned in sequence. As a consequence, the first Node / NetDevice will have “192.168.1.1”, the second “192.168.1.2” and so on.

It is possible to repeat the above to assign more than one address to a node. However, due to the Ipv4AddressHelper singleton nature, one should first assign all the addresses of a network, then change the network base (SetBase), then do a new assignment.

Alternatively, it is possible to assign a specific address to a node:

Ptr<Node> n0 = CreateObject<Node> ();
NodeContainer net (n0);
CsmaHelper csma;
NetDeviceContainer ndc = csma.Install (net);
NS_LOG_INFO ("Specifically Assign an IPv4 Address.");
Ipv4AddressHelper ipv4;
Ptr<NetDevice> device = ndc.Get (0);
Ptr<Node> node = device->GetNode ();
Ptr<Ipv4> ipv4proto = node->GetObject<Ipv4> ();
int32_t ifIndex = 0;
ifIndex = ipv4proto->GetInterfaceForDevice (device);
Ipv4InterfaceAddress ipv4Addr = Ipv4InterfaceAddress (Ipv4Address ("192.168.1.42"), NetMask ("/24"));
ipv4proto->AddAddress (ifIndex, ipv4Addr);

DHCP assigned IPv4 addresses¶

DHCP is available in the internet-apps module. In order to use DHCP you have to have a DhcpServer application in a node (the DHC server node) and a DhcpClient application in each of the nodes. Note that it not necessary that all the nodes in a subnet use DHCP. Some nodes can have static addresses.

All the DHCP setup is performed though the DhcpHelper class. A complete example is in src/internet-apps/examples/dhcp-example.cc.

Further info about the DHCP functionalities can be found in the internet-apps model documentation.

Tracing in the IPv4 Stack¶

The internet stack provides a number of trace sources in its various protocol implementations. These trace sources can be hooked using your own custom trace code, or you can use our helper functions in some cases to arrange for tracing to be enabled.

Tracing in ARP¶

ARP provides two trace hooks, one in the cache, and one in the layer three protocol. The trace accessor in the cache is given the name “Drop.” When a packet is transmitted over an interface that requires ARP, it is first queued for transmission in the ARP cache until the required MAC address is resolved. There are a number of retries that may be done trying to get the address, and if the maximum retry count is exceeded the packet in question is dropped by ARP. The single trace hook in the ARP cache is called,

A second trace hook lives in the ARP L3 protocol (also named “Drop”) and may be called for a number of reasons.

Tracing in IPv4¶

The IPv4 layer three protocol provides three trace hooks. These are the “Tx” (ns3::Ipv4L3Protocol::m_txTrace), “Rx” (ns3::Ipv4L3Protocol::m_rxTrace) and “Drop” (ns3::Ipv4L3Protocol::m_dropTrace) trace sources.

The “Tx” trace is fired in a number of situations, all of which indicate that a given packet is about to be sent down to a given ns3::Ipv4Interface.

The “Rx” trace is fired when a packet is passed from the device up to the ns3::Ipv4L3Protocol::Receive function.

The “Drop” trace is fired in any case where the packet is dropped (in both the transmit and receive paths).

Explicit Congestion Notification (ECN) bits¶

Ipv4QueueDiscItem¶

The traffic control sublayer in ns-3 handles objects of class QueueDiscItem which are used to hold an ns3::Packet and an ns3::Header. This is done to facilitate the marking of packets for Explicit Congestion Notification. The Mark () method is implemented in Ipv4QueueDiscItem. It returns true if marking the packet is successful, i.e., it successfully sets the CE bit in the IPv4 header. The Mark () method will return false, however, if the IPv4 header indicates the ECN_NotECT codepoint.

RFC 6621 duplicate packet detection¶

To support mesh network protocols over broadcast-capable networks (e.g. Wi-Fi), it is useful to have support for duplicate packet detection and filtering, since nodes in a network may receive multiple copies of flooded multicast packets arriving on different paths. The Ipv4L3Protocol model in ns-3 has a model for hash-based duplicate packet detection (DPD) based on Section 6.2.2 of (RFC 6621). The model, disabled by default, must be enabled by setting EnableRFC6621 to true. A second attribute, DuplicateExpire, sets the expiration delay for erasing the cache entry of a packet in the duplicate cache; the delay value defaults to 1ms.

IPv6¶

This chapter describes the ns-3 IPv6 model capabilities and limitations along with its usage and examples.

IPv6 model description¶

The IPv6 model is loosely patterned after the Linux implementation; the implementation is not complete as some features of IPv6 are not of much interest to simulation studies, and some features of IPv6 are simply not modeled yet in ns-3.

The base class Ipv6 defines a generic API, while the class Ipv6L3Protocol is the actual class implementing the protocol. The actual classes used by the IPv6 stack are located mainly in the directory src/internet.

The implementation of IPv6 is contained in the following files:

src/internet/model/icmpv6-header.{cc,h}
src/internet/model/icmpv6-l4-protocol.{cc,h}
src/internet/model/ipv6.{cc,h}
src/internet/model/ipv6-address-generator.{cc,h}
src/internet/model/ipv6-autoconfigured-prefix.{cc,h}
src/internet/model/ipv6-end-point.{cc,h}
src/internet/model/ipv6-end-point-demux.{cc,h}
src/internet/model/ipv6-extension.{cc,h}
src/internet/model/ipv6-extension-demux.{cc,h}
src/internet/model/ipv6-extension-header.{cc,h}
src/internet/model/ipv6-header.{cc,h}
src/internet/model/ipv6-interface.{cc,h}
src/internet/model/ipv6-interface-address.{cc,h}
src/internet/model/ipv6-l3-protocol.{cc,h}
src/internet/model/ipv6-list-routing.{cc,h}
src/internet/model/ipv6-option.{cc,h}
src/internet/model/ipv6-option-demux.{cc,h}
src/internet/model/ipv6-option-header.{cc,h}
src/internet/model/ipv6-packet-info-tag.{cc,h}
src/internet/model/ipv6-pmtu-cache.{cc,h}
src/internet/model/ipv6-raw-socket-factory.{cc,h}
src/internet/model/ipv6-raw-socket-factory-impl.{cc,h}
src/internet/model/ipv6-raw-socket-impl.{cc,h}
src/internet/model/ipv6-route.{cc,h}
src/internet/model/ipv6-routing-protocol.{cc,h}
src/internet/model/ipv6-routing-table-entry.{cc,h}
src/internet/model/ipv6-static-routing.{cc,h}
src/internet/model/ndisc-cache.{cc,h}
src/network/utils/inet6-socket-address.{cc,h}
src/network/utils/ipv6-address.{cc,h}

Also some helpers are involved with IPv6:

src/internet/helper/internet-stack-helper.{cc,h}
src/internet/helper/ipv6-address-helper.{cc,h}
src/internet/helper/ipv6-interface-container.{cc,h}
src/internet/helper/ipv6-list-routing-helper.{cc,h}
src/internet/helper/ipv6-routing-helper.{cc,h}
src/internet/helper/ipv6-static-routing-helper.{cc,h}

The model files can be roughly divided into:

Usage¶

The following description is based on using the typical helpers found in the example code.

IPv6 does not need to be activated in a node. it is automatically added to the available protocols once the Internet Stack is installed.

In order to not install IPv6 along with IPv4, it is possible to use ns3::InternetStackHelper method SetIpv6StackInstall (bool enable) before installing the InternetStack in the nodes.

Note that to have an IPv6-only network (i.e., to not install the IPv4 stack in a node) one should use ns3::InternetStackHelper method SetIpv4StackInstall (bool enable) before the stack installation.

As an example, in the following code node 0 will have both IPv4 and IPv6, node 1 only only IPv6 and node 2 only IPv4:

NodeContainer n;
n.Create (3);
InternetStackHelper internet;
InternetStackHelper internetV4only;
InternetStackHelper internetV6only;
internetV4only.SetIpv6StackInstall (false);
internetV6only.SetIpv4StackInstall (false);
internet.Install (n.Get (0));
internetV6only.Install (n.Get (1));
internetV4only.Install (n.Get (2));

IPv6 addresses assignment¶

In order to use IPv6 on a network, the first thing to do is assigning IPv6 addresses.

Any IPv6-enabled ns-3 node will have at least one NetDevice: the ns3::LoopbackNetDevice. The loopback device address is ::1. All the other NetDevices will have one or more IPv6 addresses:

Typically the first address on an interface will be the link-local one, with the global address(es) being the following ones.

IPv6 global addresses might be:

ns-3 can use both methods, and it’s quite important to understand the implications of both.

Manually assigned IPv6 addresses¶

This is probably the easiest and most used method. As an example:

Ptr<Node> n0 = CreateObject<Node> ();
Ptr<Node> n1 = CreateObject<Node> ();
NodeContainer net (n0, n1);
CsmaHelper csma;
NetDeviceContainer ndc = csma.Install (net);
NS_LOG_INFO ("Assign IPv6 Addresses.");
Ipv6AddressHelper ipv6;
ipv6.SetBase (Ipv6Address ("2001:db8::"), Ipv6Prefix (64));
Ipv6InterfaceContainer ic = ipv6.Assign (ndc);

This method will add two global IPv6 addresses to the nodes. Note that, as usual for IPv6, all the nodes will also have a link-local address. Typically the first address on an interface will be the link-local one, with the global address(es) being the following ones.

Note that the global addresses will be derived from the MAC address. As a consequence, expect to have addresses similar to 2001:db8::200:ff:fe00:1.

It is possible to repeat the above to assign more than one global address to a node. However, due to the Ipv6AddressHelper singleton nature, one should first assign all the addresses of a network, then change the network base (SetBase), then do a new assignment.

Alternatively, it is possible to assign a specific address to a node:

Ptr<Node> n0 = CreateObject<Node> ();
NodeContainer net (n0);
CsmaHelper csma;
NetDeviceContainer ndc = csma.Install (net);
NS_LOG_INFO ("Specifically Assign an IPv6 Address.");
Ipv6AddressHelper ipv6;
Ptr<NetDevice> device = ndc.Get (0);
Ptr<Node> node = device->GetNode ();
Ptr<Ipv6> ipv6proto = node->GetObject<Ipv6> ();
int32_t ifIndex = 0;
ifIndex = ipv6proto->GetInterfaceForDevice (device);
Ipv6InterfaceAddress ipv6Addr = Ipv6InterfaceAddress (Ipv6Address ("2001:db8:f00d:cafe::42"), Ipv6Prefix (64));
ipv6proto->AddAddress (ifIndex, ipv6Addr);

Auto-generated IPv6 addresses¶

This is accomplished by relying on the RADVD protocol, implemented by the class Radvd. A helper class is available, which can be used to ease the most common tasks, e.g., setting up a prefix on an interface, if it is announced periodically, and if the router is the default router for that interface.

A fine grain configuration is possible though the RadvdInterface class, which allows to setup every parameter of the announced router advertisement on a given interface.

It is worth mentioning that the configurations must be set up before installing the application in the node.

Upon using this method, the nodes will acquire dynamically (i.e., during the simulation) one (or more) global address(es) according to the RADVD configuration. These addresses will be bases on the RADVD announced prefix and the node’s EUI-64.

Examples of RADVD use are shown in examples/ipv6/radvd.cc and examples/ipv6/radvd-two-prefix.cc.

Note that the router (i.e., the node with Radvd) will have to have a global address, while the nodes using the auto-generated addresses (SLAAC) will have to have a link-local address. This is accomplished using Ipv6AddressHelper::AssignWithoutAddress, e.g.:

Ipv6AddressHelper ipv6;
NetDeviceContainer tmp;
tmp.Add (d1.Get (0)); /* n0 */
Ipv6InterfaceContainer iic1 = ipv6.AssignWithoutAddress (tmp); /* n0 interface */

Random-generated IPv6 addresses¶

While IPv6 real nodes will use randomly generated addresses to protect privacy, ns-3 does NOT have this capability. This feature haven’t been so far considered as interesting for simulation.

Networks with and without the onlink property¶

IPv6 adds the concept of “on-link” for addresses and prefixes. Simplifying the concept, a network with the on-link property behaves roughly as IPv4: a node will assume that any address belonging to the on-link prefix is reachable on the link, so it uses Neighbor Discovery Protocol (NDP) to find the MAC address corresponding to the IPv6 address. If the prefix is not marked as “on-link”, then any packet is sent to the default router.

Radvd can announce prefixes that have the on-link flag not set. Moreover, it is possible to add an address to a node without setting the on-link property for the prefix used in the address. The function to use is Ipv6AddressHelper::AssignWithoutOnLink.

Duplicate Address Detection (DAD)¶

Nodes will perform DAD (it can be disabled using an Icmpv6L4Protocol attribute). Upon receiving a DAD, however, nodes will not react to it. As is: DAD reaction is incomplete so far. The main reason relies on the missing random-generated address capability. Moreover, since ns-3 nodes will usually be well-behaving, there shouldn’t be any Duplicate Address. This might be changed in the future, so as to avoid issues with real-world integrated simulations.

Explicit Congestion Notification (ECN) bits in IPv6¶

Ipv6QueueDiscItem¶

The traffic control sublayer in ns-3 handles objects of class QueueDiscItem which are used to hold an ns3::Packet and an ns3::Header. This is done to facilitate the marking of packets for Explicit Congestion Notification. The Mark () method is implemented in Ipv6QueueDiscItem. It returns true if marking the packet is successful, i.e., it successfully sets the CE bit in the IPv6 header. The Mark () method will return false, however, if the IPv6 header indicates the ECN_NotECT codepoint.

Host and Router behaviour in IPv6 and ns-3¶

In IPv6 there is a clear distinction between routers and hosts. As one might expect, routers can forward packets from an interface to another interface, while hosts drop packets not directed to them.

Unfortunately, forwarding is not the only thing affected by this distinction, and forwarding itself might be fine-tuned, e.g., to forward packets incoming from an interface and drop packets from another interface.

In ns-3 a node is configured to be an host by default. There are two main ways to change this behaviour:

Either one can be used during the simulation.

A fine-grained setup can be accomplished by using ns3::Ipv6Interface SetForwarding (bool forward); which allows to change the behaviour on a per-interface-basis.

Note that the node-wide configuration only serves as a convenient method to enable/disable the ns3::Ipv6Interface specific setting. An Ipv6Interface added to a node with forwarding enabled will be set to be forwarding as well. This is really important when a node has interfaces added during the simulation.

According to the ns3::Ipv6Interface forwarding state, the following happens:

The behaviour is matching ip-sysctl.txt (http://www.kernel.org/doc/Documentation/networking/ip-sysctl.txt) with the difference that it’s not possible to override the behaviour using esoteric settings (e.g., forwarding but accept router advertisements, accept_ra=2, or forwarding but send router solicitations forwarding=2).

Consider carefully the implications of packet forwarding. As an example, a node will NOT send ICMPv6 PACKET_TOO_BIG messages from an interface with forwarding off. This is completely normal, as the Routing protocol will drop the packet before attempting to forward it.

Helpers¶

Typically the helpers used in IPv6 setup are:

The use is almost identical to the corresponding IPv4 case, e.g.:

NodeContainer n;
n.Create (4);
NS_LOG_INFO ("Create IPv6 Internet Stack");
InternetStackHelper internetv6;
internetv6.Install (n);
NS_LOG_INFO ("Create channels.");
CsmaHelper csma;
NetDeviceContainer d = csma.Install (n);
NS_LOG_INFO ("Create networks and assign IPv6 Addresses.");
Ipv6AddressHelper ipv6;
ipv6.SetBase (Ipv6Address ("2001:db8::"), Ipv6Prefix (64));
Ipv6InterfaceContainer iic = ipv6.Assign (d);

Additionally, a common task is to enable forwarding on one of the nodes and to setup a default route toward it in the other nodes, e.g.:

iic.SetForwarding (0, true);
iic.SetDefaultRouteInAllNodes (0);

This will enable forwarding on the node 0 and will setup a default route in ns3::Ipv6StaticRouting on all the other nodes. Note that this requires that Ipv6StaticRouting is present in the nodes.

The IPv6 routing helpers enable the user to perform specific tasks on the particular routing algorithm and to print the routing tables.

Attributes¶

Many classes in the ns-3 IPv6 implementation contain attributes. The most useful ones are:

Output¶

The IPv6 stack provides some useful trace sources:

The latest trace source is generated when a packet contains an unknown option blocking its processing.

Mind that ns3::NdiscCache could drop packets as well, and they are not logged in a trace source (yet). This might generate some confusion in the sent/received packets counters.

Advanced Usage¶

IPv6 maximum transmission unit (MTU) and fragmentation¶

ns-3 NetDevices define the MTU according to the L2 simulated Device. IPv6 requires that the minimum MTU is 1280 bytes, so all NetDevices are required to support at least this MTU. This is the link-MTU.

In order to support different MTUs in a source-destination path, ns-3 IPv6 model can perform fragmentation. This can be either triggered by receiving a packet bigger than the link-MTU from the L4 protocols (UDP, TCP, etc.), or by receiving an ICMPv6 PACKET_TOO_BIG message. The model mimics RFC 1981, with the following notable exceptions:

Both limitations are going to be removed in due time.

The Path-MTU cache is currently based on the source-destination IPv6 addresses. Further classifications (e.g., flow label) are possible but not yet implemented.

The Path-MTU default validity time is 10 minutes. After the cache entry expiration, the Path-MTU information is removed and the next packet will (eventually) trigger a new ICMPv6 PACKET_TOO_BIG message. Note that 1) this is consistent with the RFC specification and 2) L4 protocols are responsible for retransmitting the packets.

Examples¶

The examples for IPv6 are in the directory examples/ipv6. These examples focus on the most interesting IPv6 peculiarities, such as fragmentation, redirect and so on.

Moreover, most TCP and UDP examples located in examples/udp, examples/tcp, etc. have a command-line option to use IPv6 instead of IPv4.

Troubleshooting¶

There are just a few pitfalls to avoid while using ns-3 IPv6.

Routing loops¶

Since the only (so far) routing scheme available for IPv6 is ns3::Ipv6StaticRouting, default router have to be setup manually. When there are two or more routers in a network (e.g., node A and node B), avoid using the helper function SetDefaultRouteInAllNodes for more than one router.

The consequence would be to install a default route to B in A and a default route pointing to A in B, generating a loop.

Global address leakage¶

Remember that addresses in IPv6 are global by definition. When using IPv6 with an emulation ns-3 capability, avoid at all costs address leakage toward the global Internet. It is advisable to setup an external firewall to prevent leakage.

2001:DB8::/32 addresses¶

IPv6 standard (RFC 3849) defines the 2001:DB8::/32 address class for the documentation. This manual uses this convention. The addresses in this class are, however, only usable in a document, and routers should discard them.

Validation¶

The IPv6 protocols has not yet been extensively validated against real implementations. The actual tests involve mainly performing checks of the .pcap trace files with Wireshark, and the results are positive.

Routing overview¶

ns-3 is intended to support traditional routing approaches and protocols, support ports of open source routing implementations, and facilitate research into unorthodox routing techniques. The overall routing architecture is described below in Routing architecture. Users who wish to just read about how to configure global routing for wired topologies can read Global centralized routing. Unicast routing protocols are described in Unicast routing. Multicast routing is documented in Multicast routing.

Routing architecture¶

Unicast routing¶

The following unicast routing protocols are defined for IPv4 and IPv6:

In the future, this architecture should also allow someone to implement a Linux-like implementation with routing cache, or a Click modular router, but those are out of scope for now.

Ipv[4,6]ListRouting¶

This section describes the current default ns-3 Ipv[4,6]RoutingProtocol. Typically, multiple routing protocols are supported in user space and coordinate to write a single forwarding table in the kernel. Presently in ns-3, the implementation instead allows for multiple routing protocols to build/keep their own routing state, and the IP implementation will query each one of these routing protocols (in some order determined by the simulation author) until a route is found.

We chose this approach because it may better facilitate the integration of disparate routing approaches that may be difficult to coordinate the writing to a single table, approaches where more information than destination IP address (e.g., source routing) is used to determine the next hop, and on-demand routing approaches where packets must be cached.

Ipv[4,6]ListRouting::AddRoutingProtocol¶

Classes Ipv4ListRouting and Ipv6ListRouting provides a pure virtual function declaration for the method that allows one to add a routing protocol:

void AddRoutingProtocol (Ptr<Ipv4RoutingProtocol> routingProtocol,


                         int16_t priority);
void AddRoutingProtocol (Ptr<Ipv6RoutingProtocol> routingProtocol,


                         int16_t priority);

These methods are implemented respectively by class Ipv4ListRoutingImpl and by class Ipv6ListRoutingImpl in the internet module.

The priority variable above governs the priority in which the routing protocols are inserted. Notice that it is a signed int. By default in ns-3, the helper classes will instantiate a Ipv[4,6]ListRoutingImpl object, and add to it an Ipv[4,6]StaticRoutingImpl object at priority zero. Internally, a list of Ipv[4,6]RoutingProtocols is stored, and and the routing protocols are each consulted in decreasing order of priority to see whether a match is found. Therefore, if you want your Ipv4RoutingProtocol to have priority lower than the static routing, insert it with priority less than 0; e.g.:

Ptr<MyRoutingProtocol> myRoutingProto = CreateObject<MyRoutingProtocol> ();
listRoutingPtr->AddRoutingProtocol (myRoutingProto, -10);

Upon calls to RouteOutput() or RouteInput(), the list routing object will search the list of routing protocols, in priority order, until a route is found. Such routing protocol will invoke the appropriate callback and no further routing protocols will be searched.

Global centralized routing¶

Global centralized routing is sometimes called “God” routing; it is a special implementation that walks the simulation topology and runs a shortest path algorithm, and populates each node’s routing tables. No actual protocol overhead (on the simulated links) is incurred with this approach. It does have a few constraints:

Presently, global centralized IPv4 unicast routing over both point-to-point and shared (CSMA) links is supported.

By default, when using the ns-3 helper API and the default InternetStackHelper, global routing capability will be added to the node, and global routing will be inserted as a routing protocol with lower priority than the static routes (i.e., users can insert routes via Ipv4StaticRouting API and they will take precedence over routes found by global routing).

Global Unicast Routing API¶

The public API is very minimal. User scripts include the following:

#include "ns3/internet-module.h"

If the default InternetStackHelper is used, then an instance of global routing will be aggregated to each node. After IP addresses are configured, the following function call will cause all of the nodes that have an Ipv4 interface to receive forwarding tables entered automatically by the GlobalRouteManager:

Ipv4GlobalRoutingHelper::PopulateRoutingTables ();

Note: A reminder that the wifi NetDevice will work but does not take any wireless effects into account. For wireless, we recommend OLSR dynamic routing described below.

It is possible to call this function again in the midst of a simulation using the following additional public function:

Ipv4GlobalRoutingHelper::RecomputeRoutingTables ();

which flushes the old tables, queries the nodes for new interface information, and rebuilds the routes.

For instance, this scheduling call will cause the tables to be rebuilt at time 5 seconds:

Simulator::Schedule (Seconds (5),


                     &Ipv4GlobalRoutingHelper::RecomputeRoutingTables);

There are two attributes that govern the behavior. The first is Ipv4GlobalRouting::RandomEcmpRouting. If set to true, packets are randomly routed across equal-cost multipath routes. If set to false (default), only one route is consistently used. The second is Ipv4GlobalRouting::RespondToInterfaceEvents. If set to true, dynamically recompute the global routes upon Interface notification events (up/down, or add/remove address). If set to false (default), routing may break unless the user manually calls RecomputeRoutingTables() after such events. The default is set to false to preserve legacy ns-3 program behavior.

Global Routing Implementation¶

This section is for those readers who care about how this is implemented. A singleton object (GlobalRouteManager) is responsible for populating the static routes on each node, using the public Ipv4 API of that node. It queries each node in the topology for a “globalRouter” interface. If found, it uses the API of that interface to obtain a “link state advertisement (LSA)” for the router. Link State Advertisements are used in OSPF routing, and we follow their formatting.

It is important to note that all of these computations are done before packets are flowing in the network. In particular, there are no overhead or control packets being exchanged when using this implementation. Instead, this global route manager just walks the list of nodes to build the necessary information and configure each node’s routing table.

The GlobalRouteManager populates a link state database with LSAs gathered from the entire topology. Then, for each router in the topology, the GlobalRouteManager executes the OSPF shortest path first (SPF) computation on the database, and populates the routing tables on each node.

The quagga (http://www.quagga.net) OSPF implementation was used as the basis for the routing computation logic. One benefit of following an existing OSPF SPF implementation is that OSPF already has defined link state advertisements for all common types of network links:

Therefore, we think that enabling these other link types will be more straightforward now that the underlying OSPF SPF framework is in place.

Presently, we can handle IPv4 point-to-point, numbered links, as well as shared broadcast (CSMA) links. Equal-cost multipath is also supported. Although wireless link types are supported by the implementation, note that due to the nature of this implementation, any channel effects will not be considered and the routing tables will assume that every node on the same shared channel is reachable from every other node (i.e. it will be treated like a broadcast CSMA link).

The GlobalRouteManager first walks the list of nodes and aggregates a GlobalRouter interface to each one as follows:

typedef std::vector < Ptr<Node> >::iterator Iterator;
for (Iterator i = NodeList::Begin (); i != NodeList::End (); i++)


  {


    Ptr<Node> node = *i;


    Ptr<GlobalRouter> globalRouter = CreateObject<GlobalRouter> (node);


    node->AggregateObject (globalRouter);


  }

This interface is later queried and used to generate a Link State Advertisement for each router, and this link state database is fed into the OSPF shortest path computation logic. The Ipv4 API is finally used to populate the routes themselves.

RIP and RIPng¶

The RIPv2 protocol for IPv4 is described in the RFC 2453, and it consolidates a number of improvements over the base protocol defined in RFC 1058.

This IPv6 routing protocol (RFC 2080) is the evolution of the well-known RIPv1 (see RFC 1058 and RFC 1723) routing protocol for IPv4.

The protocols are very simple, and are normally suitable for flat, simple network topologies.

RIPv1, RIPv2, and RIPng have the very same goals and limitations. In particular, RIP considers any route with a metric equal or greater than 16 as unreachable. As a consequence, the maximum number of hops is the network must be less than 15 (the number of routers is not set). Users are encouraged to read RFC 2080 and RFC 1058 to fully understand RIP behaviour and limitations.

Routing convergence¶

RIP uses a Distance-Vector algorithm, and routes are updated according to the Bellman-Ford algorithm (sometimes known as Ford-Fulkerson algorithm). The algorithm has a convergence time of O(|V|*|E|) where |V| and |E| are the number of vertices (routers) and edges (links) respectively. It should be stressed that the convergence time is the number of steps in the algorithm, and each step is triggered by a message. Since Triggered Updates (i.e., when a route is changed) have a 1-5 seconds cooldown, the topology can require some time to be stabilized.

Users should be aware that, during routing tables construction, the routers might drop packets. Data traffic should be sent only after a time long enough to allow RIP to build the network topology. Usually 80 seconds should be enough to have a suboptimal (but working) routing setup. This includes the time needed to propagate the routes to the most distant router (16 hops) with Triggered Updates.

If the network topology is changed (e.g., a link is broken), the recovery time might be quite high, and it might be even higher than the initial setup time. Moreover, the network topology recovery is affected by the Split Horizoning strategy.

The examples examples/routing/ripng-simple-network.cc and examples/routing/rip-simple-network.cc shows both the network setup and network recovery phases.

Split Horizoning¶

Split Horizon is a strategy to prevent routing instability. Three options are possible:

In the first case, routes are advertised on all the router’s interfaces. In the second case, routers will not advertise a route on the interface from which it was learned. Poison Reverse will advertise the route on the interface from which it was learned, but with a metric of 16 (infinity). For a full analysis of the three techniques, see RFC 1058, section 2.2.

The examples are based on the network topology described in the RFC, but it does not show the effect described there.

The reason are the Triggered Updates, together with the fact that when a router invalidates a route, it will immediately propagate the route unreachability, thus preventing most of the issues described in the RFC.

However, with complex topologies, it is still possible to have route instability phenomena similar to the one described in the RFC after a link failure. As a consequence, all the considerations about Split Horizon remains valid.

Default routes¶

RIP protocol should be installed only on routers. As a consequence, nodes will not know what is the default router.

To overcome this limitation, users should either install the default route manually (e.g., by resorting to Ipv4StaticRouting or Ipv6StaticRouting), or by using RADVd (in case of IPv6). RADVd is available in ns-3 in the Applications module, and it is strongly suggested.

Protocol parameters and options¶

The RIP ns-3 implementations allow to change all the timers associated with route updates and routes lifetime.

Moreover, users can change the interface metrics on a per-node basis.

The type of Split Horizoning (to avoid routes back-propagation) can be selected on a per-node basis, with the choices being “no split horizon”, “split horizon” and “poison reverse”. See RFC 2080 for further details, and RFC 1058 for a complete discussion on the split horizoning strategies.

Moreover, it is possible to use a non-standard value for Link Down Value (i.e., the value after which a link is considered down). The default is value is 16.

Limitations¶

There is no support for the Next Hop option (RFC 2080, Section 2.1.1). The Next Hop option is useful when RIP is not being run on all of the routers on a network. Support for this option may be considered in the future.

There is no support for CIDR prefix aggregation. As a result, both routing tables and route advertisements may be larger than necessary. Prefix aggregation may be added in the future.

Other routing protocols¶

Other routing protocols documentation can be found under the respective modules sections, e.g.:

Multicast routing¶

The following function is used to add a static multicast route to a node:

void
Ipv4StaticRouting::AddMulticastRoute (Ipv4Address origin,


                                      Ipv4Address group,


                                      uint32_t inputInterface,


                                      std::vector<uint32_t> outputInterfaces);

A multicast route must specify an origin IP address, a multicast group and an input network interface index as conditions and provide a vector of output network interface indices over which packets matching the conditions are sent.

Typically there are two main types of multicast routes:

In the first case all the conditions must be explicitly provided.

In the second case, the route is equivalent to a unicast route, and must be added through Ipv4StaticRouting::AddHostRouteTo.

Another command sets the default multicast route:

void
Ipv4StaticRouting::SetDefaultMulticastRoute (uint32_t outputInterface);

This is the multicast equivalent of the unicast version SetDefaultRoute. We tell the routing system what to do in the case where a specific route to a destination multicast group is not found. The system forwards packets out the specified interface in the hope that “something out there” knows better how to route the packet. This method is only used in initially sending packets off of a host. The default multicast route is not consulted during forwarding – exact routes must be specified using AddMulticastRoute for that case.

Since we’re basically sending packets to some entity we think may know better what to do, we don’t pay attention to “subtleties” like origin address, nor do we worry about forwarding out multiple interfaces. If the default multicast route is set, it is returned as the selected route from LookupStatic irrespective of origin or multicast group if another specific route is not found.

Finally, a number of additional functions are provided to fetch and remove multicast routes:

uint32_t GetNMulticastRoutes (void) const;
Ipv4MulticastRoute *GetMulticastRoute (uint32_t i) const;
Ipv4MulticastRoute *GetDefaultMulticastRoute (void) const;
bool RemoveMulticastRoute (Ipv4Address origin,


                           Ipv4Address group,


                           uint32_t inputInterface);
void RemoveMulticastRoute (uint32_t index);

TCP models in ns-3¶

This chapter describes the TCP models available in ns-3.

Overview of support for TCP¶

ns-3 was written to support multiple TCP implementations. The implementations inherit from a few common header classes in the src/network directory, so that user code can swap out implementations with minimal changes to the scripts.

There are three important abstract base classes:

There are presently two active and one legacy implementations of TCP available for ns-3.

NSC is no longer actively supported; it requires use of gcc-5 or gcc-4.9, and only covers up to Linux kernel version 2.6.29.

Direct Code Execution is also limited in its support for newer kernels; at present, only Linux kernel 4.4 is supported. However, the TCP implementations in kernel 4.4 can still be used for ns-3 validation or for specialized simulation use cases.

It should also be mentioned that various ways of combining virtual machines with ns-3 makes available also some additional TCP implementations, but those are out of scope for this chapter.

ns-3 TCP¶

In brief, the native ns-3 TCP model supports a full bidirectional TCP with connection setup and close logic. Several congestion control algorithms are supported, with CUBIC the default, and NewReno, Westwood, Hybla, HighSpeed, Vegas, Scalable, Veno, Binary Increase Congestion Control (BIC), Yet Another HighSpeed TCP (YeAH), Illinois, H-TCP, Low Extra Delay Background Transport (LEDBAT), TCP Low Priority (TCP-LP), Data Center TCP (DCTCP) and Bottleneck Bandwidth and RTT (BBR) also supported. The model also supports Selective Acknowledgements (SACK), Proportional Rate Reduction (PRR) and Explicit Congestion Notification (ECN). Multipath-TCP is not yet supported in the ns-3 releases.

Model history¶

Until the ns-3.10 release, ns-3 contained a port of the TCP model from GTNetS, developed initially by George Riley and ported to ns-3 by Raj Bhattacharjea. This implementation was substantially rewritten by Adriam Tam for ns-3.10. In 2015, the TCP module was redesigned in order to create a better environment for creating and carrying out automated tests. One of the main changes involves congestion control algorithms, and how they are implemented.

Before the ns-3.25 release, a congestion control was considered as a stand-alone TCP through an inheritance relation: each congestion control (e.g. TcpNewReno) was a subclass of TcpSocketBase, reimplementing some inherited methods. The architecture was redone to avoid this inheritance, by making each congestion control a separate class, and defining an interface to exchange important data between TcpSocketBase and the congestion modules. The Linux tcp_congestion_ops interface was used as the design reference.

Along with congestion control, Fast Retransmit and Fast Recovery algorithms have been modified; in previous releases, these algorithms were delegated to TcpSocketBase subclasses. Starting from ns-3.25, they have been merged inside TcpSocketBase. In future releases, they can be extracted as separate modules, following the congestion control design.

As of the ns-3.31 release, the default initial window was set to 10 segments (in previous releases, it was set to 1 segment). This aligns with current Linux default, and is discussed further in RFC 6928.

In the ns-3.32 release, the default recovery algorithm was set to Proportional Rate Reduction (PRR) from the classic ack-clocked Fast Recovery algorithm.

In the ns-3.34 release, the default congestion control algorithm was set to CUBIC from NewReno.

Acknowledgments¶

As mentioned above, ns-3 TCP has had multiple authors and maintainers over the years. Several publications exist on aspects of ns-3 TCP, and users of ns-3 TCP are requested to cite one of the applicable papers when publishing new work.

A general reference on the current architecture is found in the following paper:

For an academic peer-reviewed paper on the SACK implementation in ns-3, please refer to:

Usage¶

In many cases, usage of TCP is set at the application layer by telling the ns-3 application which kind of socket factory to use.

Using the helper functions defined in src/applications/helper and src/network/helper, here is how one would create a TCP receiver:

// Create a packet sink on the star "hub" to receive these packets
uint16_t port = 50000;
Address sinkLocalAddress(InetSocketAddress (Ipv4Address::GetAny (), port));
PacketSinkHelper sinkHelper ("ns3::TcpSocketFactory", sinkLocalAddress);
ApplicationContainer sinkApp = sinkHelper.Install (serverNode);
sinkApp.Start (Seconds (1.0));
sinkApp.Stop (Seconds (10.0));

Similarly, the below snippet configures OnOffApplication traffic source to use TCP:

// Create the OnOff applications to send TCP to the server
OnOffHelper clientHelper ("ns3::TcpSocketFactory", Address ());

The careful reader will note above that we have specified the TypeId of an abstract base class TcpSocketFactory. How does the script tell ns-3 that it wants the native ns-3 TCP vs. some other one? Well, when internet stacks are added to the node, the default TCP implementation that is aggregated to the node is the ns-3 TCP. This can be overridden as we show below when using Network Simulation Cradle. So, by default, when using the ns-3 helper API, the TCP that is aggregated to nodes with an Internet stack is the native ns-3 TCP.

To configure behavior of TCP, a number of parameters are exported through the ns-3 attribute system. These are documented in the Doxygen for class TcpSocket. For example, the maximum segment size is a settable attribute.

To set the default socket type before any internet stack-related objects are created, one may put the following statement at the top of the simulation program:

Config::SetDefault ("ns3::TcpL4Protocol::SocketType", StringValue ("ns3::TcpNewReno"));

For users who wish to have a pointer to the actual socket (so that socket operations like Bind(), setting socket options, etc. can be done on a per-socket basis), Tcp sockets can be created by using the Socket::CreateSocket() method. The TypeId passed to CreateSocket() must be of type ns3::SocketFactory, so configuring the underlying socket type must be done by twiddling the attribute associated with the underlying TcpL4Protocol object. The easiest way to get at this would be through the attribute configuration system. In the below example, the Node container “n0n1” is accessed to get the zeroth element, and a socket is created on this node:

// Create and bind the socket...
TypeId tid = TypeId::LookupByName ("ns3::TcpNewReno");
Config::Set ("/NodeList/*/$ns3::TcpL4Protocol/SocketType", TypeIdValue (tid));
Ptr<Socket> localSocket =


  Socket::CreateSocket (n0n1.Get (0), TcpSocketFactory::GetTypeId ());

Above, the “*” wild card for node number is passed to the attribute configuration system, so that all future sockets on all nodes are set to NewReno, not just on node ‘n0n1.Get (0)’. If one wants to limit it to just the specified node, one would have to do something like:

// Create and bind the socket...
TypeId tid = TypeId::LookupByName ("ns3::TcpNewReno");
std::stringstream nodeId;
nodeId << n0n1.Get (0)->GetId ();
std::string specificNode = "/NodeList/" + nodeId.str () + "/$ns3::TcpL4Protocol/SocketType";
Config::Set (specificNode, TypeIdValue (tid));
Ptr<Socket> localSocket =


  Socket::CreateSocket (n0n1.Get (0), TcpSocketFactory::GetTypeId ());

Once a TCP socket is created, one will want to follow conventional socket logic and either connect() and send() (for a TCP client) or bind(), listen(), and accept() (for a TCP server). Please note that applications usually create the sockets they use automatically, and so is not straightforward to connect directly to them using pointers. Please refer to the source code of your preferred application to discover how and when it creates the socket.

TCP Socket interaction and interface with Application layer¶

In the following there is an analysis on the public interface of the TCP socket, and how it can be used to interact with the socket itself. An analysis of the callback fired by the socket is also carried out. Please note that, for the sake of clarity, we will use the terminology “Sender” and “Receiver” to clearly divide the functionality of the socket. However, in TCP these two roles can be applied at the same time (i.e. a socket could be a sender and a receiver at the same time): our distinction does not lose generality, since the following definition can be applied to both sockets in case of full-duplex mode.

----

TCP state machine (for commodity use)

Congestion Control Algorithms¶

Here follows a list of supported TCP congestion control algorithms. For an academic paper on many of these congestion control algorithms, see http://dl.acm.org/citation.cfm?id=2756518 .

NewReno¶

NewReno algorithm introduces partial ACKs inside the well-established Reno algorithm. This and other modifications are described in RFC 6582. We have two possible congestion window increment strategy: slow start and congestion avoidance. Taken from RFC 5681:

cwnd += min (N, SMSS)

During congestion avoidance, cwnd is incremented by roughly 1 full-sized segment per round-trip time (RTT), and for each congestion event, the slow start threshold is halved.

CUBIC¶

CUBIC (class TcpCubic) is the default TCP congestion control in Linux, macOS (since 2014), and Microsoft Windows (since 2017). CUBIC has two main differences with respect to a more classic TCP congestion control such as NewReno. First, during the congestion avoidance phase, the window size grows according to a cubic function (concave, then convex) with the latter convex portion designed to allow for bandwidth probing. Second, a hybrid slow start (HyStart) algorithm uses observations of delay increases in the slow start phase of window growth to try to exit slow start before window growth causes queue overflow.

CUBIC is documented in RFC 8312, and the ns-3 implementation is based on the RFC more so than the Linux implementation, although the Linux 4.4 kernel implementation (through the Direct Code Execution environment) has been used to validate the behavior and is fairly well aligned (see below section on validation).

Linux Reno¶

TCP Linux Reno (class TcpLinuxReno) is designed to provide a Linux-like implementation of TCP NewReno. The implementation of class TcpNewReno in ns-3 follows RFC standards, and increases cwnd more conservatively than does Linux Reno. Linux Reno modifies slow start and congestion avoidance algorithms to increase cwnd based on the number of bytes being acknowledged by each arriving ACK, rather than by the number of ACKs that arrive. Another major difference in implementation is that Linux maintains the congestion window in units of segments, while the RFCs define the congestion window in units of bytes.

In slow start phase, on each incoming ACK at the TCP sender side cwnd is increased by the number of previously unacknowledged bytes ACKed by the incoming acknowledgment. In contrast, in ns-3 NewReno, cwnd is increased by one segment per acknowledgment. In standards terminology, this difference is referred to as Appropriate Byte Counting (RFC 3465); Linux follows Appropriate Byte Counting while ns-3 NewReno does not.

cwnd += segAcked * segmentSize

cwnd += segmentSize

In congestion avoidance phase, the number of bytes that have been ACKed at the TCP sender side are stored in a ‘bytes_acked’ variable in the TCP control block. When ‘bytes_acked’ becomes greater than or equal to the value of the cwnd, ‘bytes_acked’ is reduced by the value of cwnd. Next, cwnd is incremented by a full-sized segment (SMSS). In contrast, in ns-3 NewReno, cwnd is increased by (1/cwnd) with a rounding off due to type casting into int.

if (m_cWndCnt >= w)


 {


   uint32_t delta = m_cWndCnt / w;


   m_cWndCnt -= delta * w;


   tcb->m_cWnd += delta * tcb->m_segmentSize;


   NS_LOG_DEBUG ("Subtracting delta * w from m_cWndCnt " << delta * w);


 }
:label: linuxrenocongavoid

if (segmentsAcked > 0)


 {


   double adder = static_cast<double> (tcb->m_segmentSize * tcb->m_segmentSize) / tcb->m_cWnd.Get ();


   adder = std::max (1.0, adder);


   tcb->m_cWnd += static_cast<uint32_t> (adder);


   NS_LOG_INFO ("In CongAvoid, updated to cwnd " << tcb->m_cWnd <<


                " ssthresh " << tcb->m_ssThresh);


 }
:label: newrenocongavoid

So, there are two main difference between the TCP Linux Reno and TCP NewReno in ns-3: 1) In TCP Linux Reno, delayed acknowledgement configuration does not affect congestion window growth, while in TCP NewReno, delayed acknowledgments cause a slower congestion window growth. 2) In congestion avoidance phase, the arithmetic for counting the number of segments acked and deciding when to increment the cwnd is different for TCP Linux Reno and TCP NewReno.

Following graphs shows the behavior of window growth in TCP Linux Reno and TCP NewReno with delayed acknowledgement of 2 segments:

HighSpeed¶

TCP HighSpeed is designed for high-capacity channels or, in general, for TCP connections with large congestion windows. Conceptually, with respect to the standard TCP, HighSpeed makes the cWnd grow faster during the probing phases and accelerates the cWnd recovery from losses. This behavior is executed only when the window grows beyond a certain threshold, which allows TCP HighSpeed to be friendly with standard TCP in environments with heavy congestion, without introducing new dangers of congestion collapse.

Mathematically:

cWnd = cWnd + \frac{a(cWnd)}{cWnd}

The function a() is calculated using a fixed RTT the value 100 ms (the lookup table for this function is taken from RFC 3649). For each congestion event, the slow start threshold is decreased by a value that depends on the size of the slow start threshold itself. Then, the congestion window is set to such value.

cWnd = (1-b(cWnd)) \cdot cWnd

The lookup table for the function b() is taken from the same RFC. More information at: http://dl.acm.org/citation.cfm?id=2756518

Hybla¶

The key idea behind TCP Hybla is to obtain for long RTT connections the same instantaneous transmission rate of a reference TCP connection with lower RTT. With analytical steps, it is shown that this goal can be achieved by modifying the time scale, in order for the throughput to be independent from the RTT. This independence is obtained through the use of a coefficient rho.

This coefficient is used to calculate both the slow start threshold and the congestion window when in slow start and in congestion avoidance, respectively.

More information at: http://dl.acm.org/citation.cfm?id=2756518

Westwood¶

Westwood and Westwood+ employ the AIAD (Additive Increase/Adaptive Decrease)· congestion control paradigm. When a congestion episode happens,· instead of halving the cwnd, these protocols try to estimate the network’s bandwidth and use the estimated value to adjust the cwnd.· While Westwood performs the bandwidth sampling every ACK reception,· Westwood+ samples the bandwidth every RTT.

More information at: http://dl.acm.org/citation.cfm?id=381704 and http://dl.acm.org/citation.cfm?id=2512757

Vegas¶

TCP Vegas is a pure delay-based congestion control algorithm implementing a proactive scheme that tries to prevent packet drops by maintaining a small backlog at the bottleneck queue. Vegas continuously samples the RTT and computes the actual throughput a connection achieves using Equation (6) and compares it with the expected throughput calculated in Equation (7). The difference between these 2 sending rates in Equation (8) reflects the amount of extra packets being queued at the bottleneck.

actual &= \frac{cWnd}{RTT}

expected &= \frac{cWnd}{BaseRTT}

diff &= expected - actual

To avoid congestion, Vegas linearly increases/decreases its congestion window to ensure the diff value falls between the two predefined thresholds, alpha and beta. diff and another threshold, gamma, are used to determine when Vegas should change from its slow-start mode to linear increase/decrease mode. Following the implementation of Vegas in Linux, we use 2, 4, and 1 as the default values of alpha, beta, and gamma, respectively, but they can be modified through the Attribute system.

More information at: http://dx.doi.org/10.1109/49.464716

Scalable¶

Scalable improves TCP performance to better utilize the available bandwidth of a highspeed wide area network by altering NewReno congestion window adjustment algorithm. When congestion has not been detected, for each ACK received in an RTT, Scalable increases its cwnd per:

cwnd = cwnd + 0.01

Following Linux implementation of Scalable, we use 50 instead of 100 to account for delayed ACK.

On the first detection of congestion in a given RTT, cwnd is reduced based on the following equation:

cwnd = cwnd - ceil(0.125 \cdot cwnd)

More information at: http://dl.acm.org/citation.cfm?id=956989

Veno¶

TCP Veno enhances Reno algorithm for more effectively dealing with random packet loss in wireless access networks by employing Vegas’s method in estimating the backlog at the bottleneck queue to distinguish between congestive and non-congestive states.

The backlog (the number of packets accumulated at the bottleneck queue) is calculated using Equation (11):

N &= Actual \cdot (RTT - BaseRTT) \\

&= Diff \cdot BaseRTT

where:

Diff &= Expected - Actual \\

&= \frac{cWnd}{BaseRTT} - \frac{cWnd}{RTT}

Veno makes decision on cwnd modification based on the calculated N and its predefined threshold beta.

Specifically, it refines the additive increase algorithm of Reno so that the connection can stay longer in the stable state by incrementing cwnd by 1/cwnd for every other new ACK received after the available bandwidth has been fully utilized, i.e. when N exceeds beta. Otherwise, Veno increases its cwnd by 1/cwnd upon every new ACK receipt as in Reno.

In the multiplicative decrease algorithm, when Veno is in the non-congestive state, i.e. when N is less than beta, Veno decrements its cwnd by only 1/5 because the loss encountered is more likely a corruption-based loss than a congestion-based. Only when N is greater than beta, Veno halves its sending rate as in Reno.

More information at: http://dx.doi.org/10.1109/JSAC.2002.807336

BIC¶

BIC (class TcpBic) is a predecessor of TCP CUBIC. In TCP BIC the congestion control problem is viewed as a search problem. Taking as a starting point the current window value and as a target point the last maximum window value (i.e. the cWnd value just before the loss event) a binary search technique can be used to update the cWnd value at the midpoint between the two, directly or using an additive increase strategy if the distance from the current window is too large.

This way, assuming a no-loss period, the congestion window logarithmically approaches the maximum value of cWnd until the difference between it and cWnd falls below a preset threshold. After reaching such a value (or the maximum window is unknown, i.e. the binary search does not start at all) the algorithm switches to probing the new maximum window with a ‘slow start’ strategy.

If a loss occur in either these phases, the current window (before the loss) can be treated as the new maximum, and the reduced (with a multiplicative decrease factor Beta) window size can be used as the new minimum.

More information at: http://ieeexplore.ieee.org/xpl/articleDetails.jsp?arnumber=1354672

YeAH¶

YeAH-TCP (Yet Another HighSpeed TCP) is a heuristic designed to balance various requirements of a state-of-the-art congestion control algorithm:

YeAH operates between 2 modes: Fast and Slow mode. In the Fast mode when the queue occupancy is small and the network congestion level is low, YeAH increments its congestion window according to the aggressive HSTCP rule. When the number of packets in the queue grows beyond a threshold and the network congestion level is high, YeAH enters its Slow mode, acting as Reno with a decongestion algorithm. YeAH employs Vegas’ mechanism for calculating the backlog as in Equation (13). The estimation of the network congestion level is shown in Equation (14).

Q = (RTT - BaseRTT) \cdot \frac{cWnd}{RTT}

L = \frac{RTT - BaseRTT}{BaseRTT}

To ensure TCP friendliness, YeAH also implements an algorithm to detect the presence of legacy Reno flows. Upon the receipt of 3 duplicate ACKs, YeAH decreases its slow start threshold according to Equation (15) if it’s not competing with Reno flows. Otherwise, the ssthresh is halved as in Reno:

ssthresh = min(max(\frac{cWnd}{8}, Q), \frac{cWnd}{2})

More information: http://www.csc.lsu.edu/~sjpark/cs7601/4-YeAH_TCP.pdf

Illinois¶

TCP Illinois is a hybrid congestion control algorithm designed for high-speed networks. Illinois implements a Concave-AIMD (or C-AIMD) algorithm that uses packet loss as the primary congestion signal to determine the direction of window update and queueing delay as the secondary congestion signal to determine the amount of change.

The additive increase and multiplicative decrease factors (denoted as alpha and beta, respectively) are functions of the current average queueing delay da as shown in Equations (16) and (17). To improve the protocol robustness against sudden fluctuations in its delay sampling, Illinois allows the increment of alpha to alphaMax only if da stays below d1 for a some (theta) amount of time.

alpha &=
\begin{cases}
\quad alphaMax & \quad \text{if } da <= d1 \\
\quad k1 / (k2 + da) & \quad \text{otherwise} \\ \end{cases}

beta &=
\begin{cases}
\quad betaMin & \quad \text{if } da <= d2 \\
\quad k3 + k4 \, da & \quad \text{if } d2 < da < d3 \\
\quad betaMax & \quad \text{otherwise} \end{cases}

where the calculations of k1, k2, k3, and k4 are shown in the following:

k1 &= \frac{(dm - d1) \cdot alphaMin \cdot alphaMax}{alphaMax - alphaMin}

k2 &= \frac{(dm - d1) \cdot alphaMin}{alphaMax - alphaMin} - d1

k3 &= \frac{alphaMin \cdot d3 - alphaMax \cdot d2}{d3 - d2}

k4 &= \frac{alphaMax - alphaMin}{d3 - d2}

Other parameters include da (the current average queueing delay), and Ta (the average RTT, calculated as sumRtt / cntRtt in the implementation) and Tmin (baseRtt in the implementation) which is the minimum RTT ever seen. dm is the maximum (average) queueing delay, and Tmax (maxRtt in the implementation) is the maximum RTT ever seen.

da &= Ta - Tmin

dm &= Tmax - Tmin

d_i &= eta_i \cdot dm

Illinois only executes its adaptation of alpha and beta when cwnd exceeds a threshold called winThresh. Otherwise, it sets alpha and beta to the base values of 1 and 0.5, respectively.

Following the implementation of Illinois in the Linux kernel, we use the following default parameter settings:

More information: http://www.doi.org/10.1145/1190095.1190166

H-TCP¶

H-TCP has been designed for high BDP (Bandwidth-Delay Product) paths. It is a dual mode protocol. In normal conditions, it works like traditional TCP with the same rate of increment and decrement for the congestion window. However, in high BDP networks, when it finds no congestion on the path after deltal seconds, it increases the window size based on the alpha function in the following:

alpha(delta)=1+10(delta-deltal)+0.5(delta-deltal)^2

where deltal is a threshold in seconds for switching between the modes and delta is the elapsed time from the last congestion. During congestion, it reduces the window size by multiplying by beta function provided in the reference paper. The calculated throughput between the last two consecutive congestion events is considered for beta calculation.

The transport TcpHtcp can be selected in the program examples/tcp/tcp-variants-comparison.cc to perform an experiment with H-TCP, although it is useful to increase the bandwidth in this example (e.g. to 20 Mb/s) to create a higher BDP link, such as

./waf --run "tcp-variants-comparison --transport_prot=TcpHtcp --bandwidth=20Mbps --duration=10"

More information (paper): http://www.hamilton.ie/net/htcp3.pdf

More information (Internet Draft): https://tools.ietf.org/html/draft-leith-tcp-htcp-06

LEDBAT¶

Low Extra Delay Background Transport (LEDBAT) is an experimental delay-based congestion control algorithm that seeks to utilize the available bandwidth on an end-to-end path while limiting the consequent increase in queueing delay on that path. LEDBAT uses changes in one-way delay measurements to limit congestion that the flow itself induces in the network.

As a first approximation, the LEDBAT sender operates as shown below:

On receipt of an ACK:

TARGET is the maximum queueing delay that LEDBAT itself may introduce in the network, and GAIN determines the rate at which the cwnd responds to changes in queueing delay; offtarget is a normalized value representing the difference between the measured current queueing delay and the predetermined TARGET delay. offtarget can be positive or negative; consequently, cwnd increases or decreases in proportion to offtarget.

Following the recommendation of RFC 6817, the default values of the parameters are:

To enable LEDBAT on all TCP sockets, the following configuration can be used:

Config::SetDefault ("ns3::TcpL4Protocol::SocketType", TypeIdValue (TcpLedbat::GetTypeId ()));

To enable LEDBAT on a chosen TCP socket, the following configuration can be used:

Config::Set ("$ns3::NodeListPriv/NodeList/1/$ns3::TcpL4Protocol/SocketType", TypeIdValue (TcpLedbat::GetTypeId ()));

The following unit tests have been written to validate the implementation of LEDBAT:

In comparison to RFC 6817, the scope and limitations of the current LEDBAT implementation are:

More information about LEDBAT is available in RFC 6817: https://tools.ietf.org/html/rfc6817

TCP-LP¶

TCP-Low Priority (TCP-LP) is a delay based congestion control protocol in which the low priority data utilizes only the excess bandwidth available on an end-to-end path. TCP-LP uses one way delay measurements as an indicator of congestion as it does not influence cross-traffic in the reverse direction.

On receipt of an ACK:

One way delay = Receiver timestamp - Receiver timestamp echo reply
Smoothed one way delay = 7/8 * Old Smoothed one way delay + 1/8 * one way delay If smoothed one way delay > owdMin + 15 * (owdMax - owdMin) / 100
if LP_WITHIN_INF
cwnd = 1
else
cwnd = cwnd / 2
Inference timer is set

where owdMin and owdMax are the minimum and maximum one way delays experienced throughout the connection, LP_WITHIN_INF indicates if TCP-LP is in inference phase or not

More information (paper): http://cs.northwestern.edu/~akuzma/rice/doc/TCP-LP.pdf

Data Center TCP (DCTCP)¶

DCTCP, specified in RFC 8257 and implemented in Linux, is a TCP congestion control algorithm for data center networks. It leverages Explicit Congestion Notification (ECN) to provide more fine-grained congestion feedback to the end hosts, and is intended to work with routers that implement a shallow congestion marking threshold (on the order of a few milliseconds) to achieve high throughput and low latency in the datacenter. However, because DCTCP does not react in the same way to notification of congestion experienced, there are coexistence (fairness) issues between it and legacy TCP congestion controllers, which is why it is recommended to only be used in controlled networking environments such as within data centers.

DCTCP extends the Explicit Congestion Notification signal to estimate the fraction of bytes that encounter congestion, rather than simply detecting that the congestion has occurred. DCTCP then scales the congestion window based on this estimate. This approach achieves high burst tolerance, low latency, and high throughput with shallow-buffered switches.

\alpha = (1 - g) * \alpha + g * F

In the above EWMA:

For send windows in which at least one ACK was received with ECE set, the sender should respond by reducing the congestion window as follows, once for every window of data:

cwnd = cwnd * (1 - \alpha / 2)

Following the recommendation of RFC 8257, the default values of the parameters are:

g = 0.0625 (i.e., 1/16)

initial alpha (\alpha) = 1

To enable DCTCP on all TCP sockets, the following configuration can be used:

Config::SetDefault ("ns3::TcpL4Protocol::SocketType", TypeIdValue (TcpDctcp::GetTypeId ()));

To enable DCTCP on a selected node, one can set the “SocketType” attribute on the TcpL4Protocol object of that node to the TcpDctcp TypeId.

The ECN is enabled automatically when DCTCP is used, even if the user has not explicitly enabled it.

DCTCP depends on a simple queue management algorithm in routers / switches to mark packets. The current implementation of DCTCP in ns-3 can use RED with a simple configuration to achieve the behavior of desired queue management algorithm.

To configure RED router for DCTCP:

Config::SetDefault ("ns3::RedQueueDisc::UseEcn", BooleanValue (true));
Config::SetDefault ("ns3::RedQueueDisc::QW", DoubleValue (1.0));
Config::SetDefault ("ns3::RedQueueDisc::MinTh", DoubleValue (16));
Config::SetDefault ("ns3::RedQueueDisc::MaxTh", DoubleValue (16));

There is also the option, when running CoDel or FqCoDel, to enable ECN on the queue and to set the “CeThreshold” value to a low value such as 1ms. The following example uses CoDel:

Config::SetDefault ("ns3::CoDelQueueDisc::UseEcn", BooleanValue (true));
Config::SetDefault ("ns3::CoDelQueueDisc::CeThreshold", TimeValue (MilliSeconds (1)));

The following unit tests have been written to validate the implementation of DCTCP:

An example program, examples/tcp/tcp-validation.cc, can be used to experiment with DCTCP for long-running flows with different bottleneck link bandwidth, base RTTs, and queuing disciplines. A variant of this program has also been run using the ns-3 Direct Code Execution environment using DCTCP from Linux kernel 4.4, and the results were compared against ns-3 results.

An example program based on an experimental topology found in the original DCTCP SIGCOMM paper is provided in examples/tcp/dctcp-example.cc. This example uses a simple topology consisting of forty DCTCP senders and receivers and two ECN-enabled switches to examine throughput, fairness, and queue delay properties of the network.

This implementation was tested extensively against a version of DCTCP in the Linux kernel version 4.4 using the ns-3 direct code execution (DCE) environment. Some differences were noted:

More information about DCTCP is available in the RFC 8257: https://tools.ietf.org/html/rfc8257

BBR¶

BBR (class TcpBbr) is a congestion control algorithm that regulates the sending rate by deriving an estimate of the bottleneck’s available bandwidth and RTT of the path. It seeks to operate at an optimal point where sender experiences maximum delivery rate with minimum RTT. It creates a network model comprising maximum delivery rate with minimum RTT observed so far, and then estimates BDP (maximum bandwidth * minimum RTT) to control the maximum amount of inflight data. BBR controls congestion by limiting the rate at which packets are sent. It caps the cwnd to one BDP and paces out packets at a rate which is adjusted based on the latest estimate of delivery rate. BBR algorithm is agnostic to packet losses and ECN marks.

pacing_gain controls the rate of sending data and cwnd_gain controls the amount of data to send.

The following is a high level overview of BBR congestion control algorithm:

To enable BBR on all TCP sockets, the following configuration can be used:

Config::SetDefault ("ns3::TcpL4Protocol::SocketType", TypeIdValue (TcpBbr::GetTypeId ()));

To enable BBR on a chosen TCP socket, the following configuration can be used (note that an appropriate Node ID must be used instead of 1):

Config::Set ("$ns3::NodeListPriv/NodeList/1/$ns3::TcpL4Protocol/SocketType", TypeIdValue (TcpBbr::GetTypeId ()));

The ns-3 implementation of BBR is based on its Linux implementation. Linux 5.4 kernel implementation has been used to validate the behavior of ns-3 implementation of BBR (See below section on Validation).

In addition, the following unit tests have been written to validate the implementation of BBR in ns-3:

of BBR.

An example program, examples/tcp/tcp-bbr-example.cc, is provided to experiment with BBR for one long running flow. This example uses a simple topology consisting of one sender, one receiver and two routers to examine congestion window, throughput and queue control. A program similar to this has been run using the Network Stack Tester (NeST) using BBR from Linux kernel 5.4, and the results were compared against ns-3 results.

More information about BBR is available in the following Internet Draft: https://tools.ietf.org/html/draft-cardwell-iccrg-bbr-congestion-control-00

More information about Delivery Rate Estimation is in the following draft: https://tools.ietf.org/html/draft-cheng-iccrg-delivery-rate-estimation-00

For an academic peer-reviewed paper on the BBR implementation in ns-3, please refer to:

Support for Explicit Congestion Notification (ECN)¶

ECN provides end-to-end notification of network congestion without dropping packets. It uses two bits in the IP header: ECN Capable Transport (ECT bit) and Congestion Experienced (CE bit), and two bits in the TCP header: Congestion Window Reduced (CWR) and ECN Echo (ECE).

More information is available in RFC 3168: https://tools.ietf.org/html/rfc3168

The following ECN states are declared in src/internet/model/tcp-socket-state.h

typedef enum


  {


    ECN_DISABLED = 0, //!< ECN disabled traffic


    ECN_IDLE,         //!< ECN is enabled but currently there is no action pertaining to ECE or CWR to be taken


    ECN_CE_RCVD,      //!< Last packet received had CE bit set in IP header


    ECN_SENDING_ECE,  //!< Receiver sends an ACK with ECE bit set in TCP header


    ECN_ECE_RCVD,     //!< Last ACK received had ECE bit set in TCP header


    ECN_CWR_SENT      //!< Sender has reduced the congestion window, and sent a packet with CWR bit set in TCP header. This is used for tracing.


  } EcnStates_t;

Current implementation of ECN is based on RFC 3168 and is referred as Classic ECN.

The following enum represents the mode of ECN:

typedef enum


  {


    ClassicEcn,  //!< ECN functionality as described in RFC 3168.


    DctcpEcn,    //!< ECN functionality as described in RFC 8257. Note: this mode is specific to DCTCP.


  } EcnMode_t;

The following are some important ECN parameters:

// ECN parameters
EcnMode_t              m_ecnMode {ClassicEcn}; //!< ECN mode
UseEcn_t               m_useEcn {Off};         //!< Socket ECN capability

Enabling ECN¶

By default, support for ECN is disabled in TCP sockets. To enable, change the value of the attribute ns3::TcpSocketBase::UseEcn to On. Following are supported values for the same, this functionality is aligned with Linux: https://www.kernel.org/doc/Documentation/networking/ip-sysctl.txt

typedef enum


  {


    Off        = 0,   //!< Disable


    On         = 1,   //!< Enable


    AcceptOnly = 2,   //!< Enable only when the peer endpoint is ECN capable


  } UseEcn_t;

For example:

Config::SetDefault ("ns3::TcpSocketBase::UseEcn", StringValue ("On"))

ECN negotiation¶

ECN capability is negotiated during the three-way TCP handshake:

if (m_useEcn == UseEcn_t::On)


  {


    SendEmptyPacket (TcpHeader::SYN | TcpHeader::ECE | TcpHeader::CWR);


  }
else


  {


    SendEmptyPacket (TcpHeader::SYN);


  }
m_ecnState = ECN_DISABLED;

if (m_useEcn != UseEcn_t::Off && (tcpHeader.GetFlags () & (TcpHeader::CWR | TcpHeader::ECE)) == (TcpHeader::CWR | TcpHeader::ECE))


  {


    SendEmptyPacket (TcpHeader::SYN | TcpHeader::ACK |TcpHeader::ECE);


    m_ecnState = ECN_IDLE;


  }
else


  {


    SendEmptyPacket (TcpHeader::SYN | TcpHeader::ACK);


    m_ecnState = ECN_DISABLED;


  }

if (m_useEcn != UseEcn_t::Off &&  (tcpHeader.GetFlags () & (TcpHeader::CWR | TcpHeader::ECE)) == (TcpHeader::ECE))


  {


    m_ecnState = ECN_IDLE;


  }
else


  {


    m_ecnState = ECN_DISABLED;


  }

Once the ECN-negotiation is successful, the sender sends data packets with ECT bits set in the IP header.

Note: As mentioned in Section 6.1.1 of RFC 3168, ECT bits should not be set during ECN negotiation. The ECN negotiation implemented in ns-3 follows this guideline.

ECN State Transitions¶

RFC 3168 compliance¶

Based on the suggestions provided in RFC 3168, the following behavior has been implemented:

Open issues¶

The following issues are yet to be addressed:

Support for Dynamic Pacing¶

TCP pacing refers to the sender-side practice of scheduling the transmission of a burst of eligible TCP segments across a time interval such as a TCP RTT, to avoid or reduce bursts. Historically, TCP used the natural ACK clocking mechanism to pace segments, but some network paths introduce aggregation (bursts of ACKs arriving) or ACK thinning, either of which disrupts ACK clocking. Some latency-sensitive congestion controls under development (Prague, BBR) require pacing to operate effectively.

Until recently, the state of the art in Linux was to support pacing in one of two ways:

The presentation by Dumazet and Cheng at IETF 88 summarizes: https://www.ietf.org/proceedings/88/slides/slides-88-tcpm-9.pdf

The first option was most often used when offloading (TSO) was enabled and when the sch_fq scheduler was used at the traffic control (qdisc) sublayer. In this case, TCP was responsible for setting the socket pacing rate, but the qdisc sublayer would enforce it. When TSO was enabled, the kernel would break a large burst into smaller chunks, with dynamic sizing based on the pacing rate, and hand off the segments to the fq qdisc for pacing.

The second option was used if sch_fq was not enabled; TCP would be responsible for internally pacing.

In 2018, Linux switched to an Early Departure Model (EDM): https://lwn.net/Articles/766564/.

TCP pacing in Linux was added in kernel 3.12, and authors chose to allow a pacing rate of 200% against the current rate, to allow probing for optimal throughput even during slow start phase. Some refinements were added in https://git.kernel.org/pub/scm/linux/kernel/git/torvalds/linux.git/commit/?id=43e122b014c9, in which Google reported that it was better to apply a different ratio (120%) in Congestion Avoidance phase. Furthermore, authors found that after cwnd reduction, it was helpful to become more conservative and switch to the conservative ratio (120%) as soon as cwnd >= ssthresh/2, as the initial ramp up (when ssthresh is infinite) still allows doubling cwnd every other RTT. Linux also does not pace the initial window (IW), typically 10 segments in practice.

Linux has also been observed to not pace if the number of eligible segments to be sent is exactly two; they will be sent back to back. If three or more, the first two are sent immediately, and additional segments are paced at the current pacing rate.

In ns-3, the model is as follows. There is no TSO/sch_fq model; only internal pacing according to current Linux policy.

Pacing may be enabled for any TCP congestion control, and a maximum pacing rate can be set. Furthermore, dynamic pacing is enabled for all TCP variants, according to the following guidelines.

Dynamic pacing is demonstrated by the example program examples/tcp/tcp-pacing.cc.

Validation¶

The following tests are found in the src/internet/test directory. In general, TCP tests inherit from a class called TcpGeneralTest, which provides common operations to set up test scenarios involving TCP objects. For more information on how to write new tests, see the section below on Writing TCP tests.

Several tests have dependencies outside of the internet module, so they are located in a system test directory called src/test/ns3tcp. Three of these six tests involve use of the Network Simulation Cradle, and are disabled if NSC is not enabled in the build.

Several TCP validation test results can also be found in the wiki page describing this implementation.

The ns-3 implementation of TCP Linux Reno was validated against the NewReno implementation of Linux kernel 4.4.0 using ns-3 Direct Code Execution (DCE). DCE is a framework which allows the users to run kernel space protocol inside ns-3 without changing the source code.

In this validation, cwnd traces of DCE Linux reno were compared to those of ns-3 Linux Reno and NewReno for a delayed acknowledgement configuration of 1 segment (in the ns-3 implementation; Linux does not allow direct configuration of this setting). It can be observed that cwnd traces for ns-3 Linux Reno are closely overlapping with DCE reno, while for ns-3 NewReno there was deviation in the congestion avoidance phase.

Writing a new congestion control algorithm¶

Writing (or porting) a congestion control algorithms from scratch (or from other systems) is a process completely separated from the internals of TcpSocketBase.

All operations that are delegated to a congestion control are contained in the class TcpCongestionOps. It mimics the structure tcp_congestion_ops of Linux, and the following operations are defined:

virtual std::string GetName () const;
virtual uint32_t GetSsThresh (Ptr<const TcpSocketState> tcb, uint32_t bytesInFlight);
virtual void IncreaseWindow (Ptr<TcpSocketState> tcb, uint32_t segmentsAcked);
virtual void PktsAcked (Ptr<TcpSocketState> tcb, uint32_t segmentsAcked,const Time& rtt);
virtual Ptr<TcpCongestionOps> Fork ();
virtual void CwndEvent (Ptr<TcpSocketState> tcb, const TcpSocketState::TcpCaEvent_t event);

The most interesting methods to write are GetSsThresh and IncreaseWindow. The latter is called when TcpSocketBase decides that it is time to increase the congestion window. Much information is available in the Transmission Control Block, and the method should increase cWnd and/or ssThresh based on the number of segments acked.

GetSsThresh is called whenever the socket needs an updated value of the slow start threshold. This happens after a loss; congestion control algorithms are then asked to lower such value, and to return it.

PktsAcked is used in case the algorithm needs timing information (such as RTT), and it is called each time an ACK is received.

CwndEvent is used in case the algorithm needs the state of socket during different congestion window event.

TCP SACK and non-SACK¶

To avoid code duplication and the effort of maintaining two different versions of the TCP core, namely RFC 6675 (TCP-SACK) and RFC 5681 (TCP congestion control), we have merged RFC 6675 in the current code base. If the receiver supports the option, the sender bases its retransmissions over the received SACK information. However, in the absence of that option, the best it can do is to follow the RFC 5681 specification (on Fast Retransmit/Recovery) and employing NewReno modifications in case of partial ACKs.

A similar concept is used in Linux with the function tcp_add_reno_sack. Our implementation resides in the TcpTxBuffer class that implements a scoreboard through two different lists of segments. TcpSocketBase actively uses the API provided by TcpTxBuffer to query the scoreboard; please refer to the Doxygen documentation (and to in-code comments) if you want to learn more about this implementation.

For an academic peer-reviewed paper on the SACK implementation in ns-3, please refer to https://dl.acm.org/citation.cfm?id=3067666.

Loss Recovery Algorithms¶

The following loss recovery algorithms are supported in ns-3 TCP. The current default (as of ns-3.32 release) is Proportional Rate Reduction (PRR), while the default for ns-3.31 and earlier was Classic Recovery.

Classic Recovery¶

Classic Recovery refers to the combination of NewReno algorithm described in RFC 6582 along with SACK based loss recovery algorithm mentioned in RFC 6675. SACK based loss recovery is used when sender and receiver support SACK options. In the case when SACK options are disabled, the NewReno modification handles the recovery.

At the start of recovery phase the congestion window is reduced diffently for NewReno and SACK based recovery. For NewReno the reduction is done as given below:

cWnd = ssThresh

For SACK based recovery, this is done as follows:

cWnd = ssThresh + (dupAckCount * segmentSize)

While in the recovery phase, the congestion window is inflated by segmentSize on arrival of every ACK when NewReno is used. The congestion window is kept same when SACK based loss recovery is used.

Proportional Rate Reduction¶

Proportional Rate Reduction (PRR) is a loss recovery algorithm described in RFC 6937 and currently used in Linux. The design of PRR helps in avoiding excess window adjustments and aims to keep the congestion window as close as possible to ssThresh.

PRR updates the congestion window by comparing the values of bytesInFlight and ssThresh. If the value of bytesInFlight is greater than ssThresh, congestion window is updated as shown below:

sndcnt = CEIL(prrDelivered * ssThresh / RecoverFS) - prrOut

cWnd = pipe + sndcnt

where RecoverFS is the value of bytesInFlight at the start of recovery phase, prrDelivered is the total bytes delivered during recovery phase, prrOut is the total bytes sent during recovery phase and sndcnt represents the number of bytes to be sent in response to each ACK.

Otherwise, the congestion window is updated by either using Conservative Reduction Bound (CRB) or Slow Start Reduction Bound (SSRB) with SSRB being the default Reduction Bound. Each Reduction Bound calculates a maximum data sending limit. For CRB, the limit is calculated as shown below:

limit = prrDelivered - prr out

For SSRB, it is calculated as:

limit = MAX(prrDelivered - prrOut, DeliveredData) + MSS

where DeliveredData represets the total number of bytes delivered to the receiver as indicated by the current ACK and MSS is the maximum segment size.

After limit calculation, the cWnd is updated as given below:

sndcnt = MIN (ssThresh - pipe, limit)

cWnd = pipe + sndcnt

More information (paper): https://dl.acm.org/citation.cfm?id=2068832

More information (RFC): https://tools.ietf.org/html/rfc6937

Adding a new loss recovery algorithm in ns-3¶

Writing (or porting) a loss recovery algorithms from scratch (or from other systems) is a process completely separated from the internals of TcpSocketBase.

All operations that are delegated to a loss recovery are contained in the class TcpRecoveryOps and are given below:

virtual std::string GetName () const;
virtual void EnterRecovery (Ptr<const TcpSocketState> tcb, uint32_t unAckDataCount,


                            bool isSackEnabled, uint32_t dupAckCount,


                            uint32_t bytesInFlight, uint32_t lastDeliveredBytes);
virtual void DoRecovery (Ptr<const TcpSocketState> tcb, uint32_t unAckDataCount,


                         bool isSackEnabled, uint32_t dupAckCount,


                         uint32_t bytesInFlight, uint32_t lastDeliveredBytes);
virtual void ExitRecovery (Ptr<TcpSocketState> tcb, uint32_t bytesInFlight);
virtual void UpdateBytesSent (uint32_t bytesSent);
virtual Ptr<TcpRecoveryOps> Fork ();

EnterRecovery is called when packet loss is detected and recovery is triggered. While in recovery phase, each time when an ACK arrives, DoRecovery is called which performs the necessary congestion window changes as per the recovery algorithm. ExitRecovery is called just prior to exiting recovery phase in order to perform the required congestion window ajustments. UpdateBytesSent is used to keep track of bytes sent and is called whenever a data packet is sent during recovery phase.

Delivery Rate Estimation¶

Current TCP implementation measures the approximate value of the delivery rate of inflight data based on Delivery Rate Estimation.

As high level idea, keep in mind that the algorithm keeps track of 2 variables:

When a packet is transmitted, the value of delivered (d0) and deliveredStamp (t0) is stored in its respective TcpTxItem.

When an acknowledgement comes for this packet, the value of delivered and deliveredStamp is updated to d1 and t1 in the same TcpTxItem.

After processing the acknowledgement, the rate sample is calculated and then passed to a congestion avoidance algorithm:

delivery_rate = (d1 - d0)/(t1 - t0)

The implementation to estimate delivery rate is a joint work between TcpTxBuffer and TcpRateOps. For more information, please take a look at their Doxygen documentation.

The implementation follows the Internet draft (Delivery Rate Estimation): https://tools.ietf.org/html/draft-cheng-iccrg-delivery-rate-estimation-00

Current limitations¶

Writing TCP tests¶

The TCP subsystem supports automated test cases on both socket functions and congestion control algorithms. To show how to write tests for TCP, here we explain the process of creating a test case that reproduces the Bug #1571.

The bug concerns the zero window situation, which happens when the receiver cannot handle more data. In this case, it advertises a zero window, which causes the sender to pause transmission and wait for the receiver to increase the window.

The sender has a timer to periodically check the receiver’s window: however, in modern TCP implementations, when the receiver has freed a “significant” amount of data, the receiver itself sends an “active” window update, meaning that the transmission could be resumed. Nevertheless, the sender timer is still necessary because window updates can be lost.

NOTE:

We describe the writing of two test cases, covering both situations: the sender’s zero-window probing and the receiver “active” window update. Our focus will be on dealing with the reported problems, which are:

However, other things should be checked in the test:

To construct the test case, one first derives from the TcpGeneralTest class:

The code is the following:

TcpZeroWindowTest::TcpZeroWindowTest (const std::string &desc)


   : TcpGeneralTest (desc)
{
}

Then, one should define the general parameters for the TCP connection, which will be one-sided (one node is acting as SENDER, while the other is acting as RECEIVER):

We have also to define the link properties, because the above definition does not work for every combination of propagation delay and sender application behavior.

To define the properties of the environment (e.g. properties which should be set before the object creation, such as propagation delay) one next implements the method ConfigureEnvironment:

void
TcpZeroWindowTest::ConfigureEnvironment ()
{


  TcpGeneralTest::ConfigureEnvironment ();


  SetAppPktCount (20);


  SetMTU (500);


  SetTransmitStart (Seconds (2.0));


  SetPropagationDelay (MilliSeconds (50));
}

For other properties, set after the object creation, one can use ConfigureProperties (). The difference is that some values, such as initial congestion window or initial slow start threshold, are applicable only to a single instance, not to every instance we have. Usually, methods that requires an id and a value are meant to be called inside ConfigureProperties (). Please see the Doxygen documentation for an exhaustive list of the tunable properties.

void
TcpZeroWindowTest::ConfigureProperties ()
{


  TcpGeneralTest::ConfigureProperties ();


  SetInitialCwnd (SENDER, 10);
}

To see the default value for the experiment, please see the implementation of both methods inside TcpGeneralTest class.

NOTE:

To define a zero-window situation, we choose (by design) to initiate the connection with a 0-byte rx buffer. This implies that the RECEIVER, in its first SYN-ACK, advertises a zero window. This can be accomplished by implementing the method CreateReceiverSocket, setting an Rx buffer value of 0 bytes (at line 6 of the following code):

Ptr<TcpSocketMsgBase>
TcpZeroWindowTest::CreateReceiverSocket (Ptr<Node> node)
{


  Ptr<TcpSocketMsgBase> socket = TcpGeneralTest::CreateReceiverSocket (node);


  socket->SetAttribute("RcvBufSize", UintegerValue (0));


  Simulator::Schedule (Seconds (10.0),


                       &TcpZeroWindowTest::IncreaseBufSize, this);


  return socket;
}

Even so, to check the active window update, we should schedule an increase of the buffer size. We do this at line 7 and 8, scheduling the function IncreaseBufSize.

void
TcpZeroWindowTest::IncreaseBufSize ()
{


  SetRcvBufSize (RECEIVER, 2500);
}

Which utilizes the SetRcvBufSize method to edit the RxBuffer object of the RECEIVER. As said before, check the Doxygen documentation for class TcpGeneralTest to be aware of the various possibilities that it offers.

NOTE:

void
TcpGeneralTest::SetRcvBufSize (SocketWho who, uint32_t size)
{


  if (who == SENDER)


    {


      m_senderSocket->SetRcvBufSize (size);


    }


  else if (who == RECEIVER)


    {


      m_receiverSocket->SetRcvBufSize (size);


    }


  else


    {


      NS_FATAL_ERROR ("Not defined");


    }
}

Next, we can start to follow the TCP connection:

While the general structure is defined, and the connection is started, we need to define a way to check the rWnd field on the segments. To this aim, we can implement the methods Rx and Tx in the TcpGeneralTest subclass, checking each time the actions of the RECEIVER and the SENDER. These methods are defined in TcpGeneralTest, and they are attached to the Rx and Tx traces in the TcpSocketBase. One should write small tests for every detail that one wants to ensure during the connection (it will prevent the test from changing over the time, and it ensures that the behavior will stay consistent through releases). We start by ensuring that the first SYN-ACK has 0 as advertised window size:

void
TcpZeroWindowTest::Tx(const Ptr<const Packet> p, const TcpHeader &h, SocketWho who)
{


  ...


  else if (who == RECEIVER)


    {


      NS_LOG_INFO ("\tRECEIVER TX " << h << " size " << p->GetSize());


      if (h.GetFlags () & TcpHeader::SYN)


        {


          NS_TEST_ASSERT_MSG_EQ (h.GetWindowSize(), 0,


                                 "RECEIVER window size is not 0 in the SYN-ACK");


        }


    }


    ....


 }

Pratically, we are checking that every SYN packet sent by the RECEIVER has the advertised window set to 0. The same thing is done also by checking, in the Rx method, that each SYN received by SENDER has the advertised window set to 0. Thanks to the log subsystem, we can print what is happening through messages. If we run the experiment, enabling the logging, we can see the following:

./waf shell
gdb --args ./build/utils/ns3-dev-test-runner-debug --test-name=tcp-zero-window-test --stop-on-failure --fullness=QUICK --assert-on-failure --verbose
(gdb) run
0.00s TcpZeroWindowTestSuite:Tx(): 0.00      SENDER TX 49153 > 4477 [SYN] Seq=0 Ack=0 Win=32768 ns3::TcpOptionWinScale(2) ns3::TcpOptionTS(0;0) size 36
0.05s TcpZeroWindowTestSuite:Rx(): 0.05      RECEIVER RX 49153 > 4477 [SYN] Seq=0 Ack=0 Win=32768 ns3::TcpOptionWinScale(2) ns3::TcpOptionTS(0;0) ns3::TcpOptionEnd(EOL) size 0
0.05s TcpZeroWindowTestSuite:Tx(): 0.05      RECEIVER TX 4477 > 49153 [SYN|ACK] Seq=0 Ack=1 Win=0 ns3::TcpOptionWinScale(0) ns3::TcpOptionTS(50;0) size 36
0.10s TcpZeroWindowTestSuite:Rx(): 0.10      SENDER RX 4477 > 49153 [SYN|ACK] Seq=0 Ack=1 Win=0 ns3::TcpOptionWinScale(0) ns3::TcpOptionTS(50;0) ns3::TcpOptionEnd(EOL) size 0
0.10s TcpZeroWindowTestSuite:Tx(): 0.10      SENDER TX 49153 > 4477 [ACK] Seq=1 Ack=1 Win=32768 ns3::TcpOptionTS(100;50) size 32
0.15s TcpZeroWindowTestSuite:Rx(): 0.15      RECEIVER RX 49153 > 4477 [ACK] Seq=1 Ack=1 Win=32768 ns3::TcpOptionTS(100;50) ns3::TcpOptionEnd(EOL) size 0
(...)

The output is cut to show the threeway handshake. As we can see from the headers, the rWnd of RECEIVER is set to 0, and thankfully our tests are not failing. Now we need to test for the persistent timer, which should be started by the SENDER after it receives the SYN-ACK. Since the Rx method is called before any computation on the received packet, we should utilize another method, namely ProcessedAck, which is the method called after each processed ACK. In the following, we show how to check if the persistent event is running after the processing of the SYN-ACK:

void
TcpZeroWindowTest::ProcessedAck (const Ptr<const TcpSocketState> tcb,


                                 const TcpHeader& h, SocketWho who)
{


  if (who == SENDER)


    {


      if (h.GetFlags () & TcpHeader::SYN)


        {


          EventId persistentEvent = GetPersistentEvent (SENDER);


          NS_TEST_ASSERT_MSG_EQ (persistentEvent.IsRunning (), true,


                                 "Persistent event not started");


        }


    }


 }

Since we programmed the increase of the buffer size after 10 simulated seconds, we expect the persistent timer to fire before any rWnd changes. When it fires, the SENDER should send a window probe, and the receiver should reply reporting again a zero window situation. At first, we investigates on what the sender sends:

  if (Simulator::Now ().GetSeconds () <= 6.0)


    {


      NS_TEST_ASSERT_MSG_EQ (p->GetSize () - h.GetSerializedSize(), 0,


                             "Data packet sent anyway");


    }


  else if (Simulator::Now ().GetSeconds () > 6.0 &&


           Simulator::Now ().GetSeconds () <= 7.0)


    {


      NS_TEST_ASSERT_MSG_EQ (m_zeroWindowProbe, false, "Sent another probe");


      if (! m_zeroWindowProbe)


        {


          NS_TEST_ASSERT_MSG_EQ (p->GetSize () - h.GetSerializedSize(), 1,


                                 "Data packet sent instead of window probe");


          NS_TEST_ASSERT_MSG_EQ (h.GetSequenceNumber(), SequenceNumber32 (1),


                                 "Data packet sent instead of window probe");


          m_zeroWindowProbe = true;


        }


    }

We divide the events by simulated time. At line 1, we check everything that happens before the 6.0 seconds mark; for instance, that no data packets are sent, and that the state remains OPEN for both sender and receiver.

Since the persist timeout is initialized at 6 seconds (exercise left for the reader: edit the test, getting this value from the Attribute system), we need to check (line 6) between 6.0 and 7.0 simulated seconds that the probe is sent. Only one probe is allowed, and this is the reason for the check at line 11.

if (Simulator::Now ().GetSeconds () > 6.0 &&


    Simulator::Now ().GetSeconds () <= 7.0)


  {


    NS_TEST_ASSERT_MSG_EQ (h.GetSequenceNumber(), SequenceNumber32 (1),


                           "Data packet sent instead of window probe");


    NS_TEST_ASSERT_MSG_EQ (h.GetWindowSize(), 0,


                           "No zero window advertised by RECEIVER");


  }

For the RECEIVER, the interval between 6 and 7 seconds is when the zero-window segment is sent.

Other checks are redundant; the safest approach is to deny any other packet exchange between the 7 and 10 seconds mark.

else if (Simulator::Now ().GetSeconds () > 7.0 &&


         Simulator::Now ().GetSeconds () < 10.0)


  {


    NS_FATAL_ERROR ("No packets should be sent before the window update");


  }

The state checks are performed at the end of the methods, since they are valid in every condition:

NS_TEST_ASSERT_MSG_EQ (GetCongStateFrom (GetTcb(SENDER)), TcpSocketState::CA_OPEN,


                       "Sender State is not OPEN");
NS_TEST_ASSERT_MSG_EQ (GetCongStateFrom (GetTcb(RECEIVER)), TcpSocketState::CA_OPEN,


                       "Receiver State is not OPEN");

Now, the interesting part in the Tx method is to check that after the 10.0 seconds mark (when the RECEIVER sends the active window update) the value of the window should be greater than zero (and precisely, set to 2500):

else if (Simulator::Now().GetSeconds() >= 10.0)


  {


    NS_TEST_ASSERT_MSG_EQ (h.GetWindowSize(), 2500,


                           "Receiver window not updated");


  }

To be sure that the sender receives the window update, we can use the Rx method:

if (Simulator::Now().GetSeconds() >= 10.0)


  {


    NS_TEST_ASSERT_MSG_EQ (h.GetWindowSize(), 2500,


                           "Receiver window not updated");


    m_windowUpdated = true;


  }

We check every packet after the 10 seconds mark to see if it has the window updated. At line 5, we also set to true a boolean variable, to check that we effectively reach this test.

Last but not least, we implement also the NormalClose() method, to check that the connection ends with a success:

void
TcpZeroWindowTest::NormalClose (SocketWho who)
{


  if (who == SENDER)


    {


      m_senderFinished = true;


    }


  else if (who == RECEIVER)


    {


      m_receiverFinished = true;


    }
}

The method is called only if all bytes are transmitted successfully. Then, in the method FinalChecks(), we check all variables, which should be true (which indicates that we have perfectly closed the connection).

void
TcpZeroWindowTest::FinalChecks ()
{


  NS_TEST_ASSERT_MSG_EQ (m_zeroWindowProbe, true,


                         "Zero window probe not sent");


  NS_TEST_ASSERT_MSG_EQ (m_windowUpdated, true,


                         "Window has not updated during the connection");


  NS_TEST_ASSERT_MSG_EQ (m_senderFinished, true,


                         "Connection not closed successfully (SENDER)");


  NS_TEST_ASSERT_MSG_EQ (m_receiverFinished, true,


                         "Connection not closed successfully (RECEIVER)");
}

To run the test, the usual way is

./test.py -s tcp-zero-window-test
PASS: TestSuite tcp-zero-window-test
1 of 1 tests passed (1 passed, 0 skipped, 0 failed, 0 crashed, 0 valgrind errors)

To see INFO messages, use a combination of ./waf shell and gdb (really useful):

./waf shell && gdb --args ./build/utils/ns3-dev-test-runner-debug --test-name=tcp-zero-window-test --stop-on-failure --fullness=QUICK --assert-on-failure --verbose

and then, hit “Run”.

NOTE:

Network Simulation Cradle¶

The Network Simulation Cradle (NSC) is a framework for wrapping real-world network code into simulators, allowing simulation of real-world behavior at little extra cost. This work has been validated by comparing situations using a test network with the same situations in the simulator. To date, it has been shown that the NSC is able to produce extremely accurate results. NSC supports four real world stacks: FreeBSD, OpenBSD, lwIP and Linux. Emphasis has been placed on not changing any of the network stacks by hand. Not a single line of code has been changed in the network protocol implementations of any of the above four stacks. However, a custom C parser was built to programmatically change source code.

NSC has previously been ported to ns-2 and OMNeT++, and was was added to ns-3 in September 2008 (ns-3.2 release). This section describes the ns-3 port of NSC and how to use it.

NSC has been obsoleted by the Linux kernel support within Direct Code Execution (DCE). However, NSC is still available through the bake build system. NSC supports Linux kernels 2.6.18 and 2.6.26, and an experimental version of 2.6.29 exists on ns-3’s code server (http://code.nsnam.org/fw/nsc-linux-2.6.29/), but newer versions of the kernel have not been ported.

Prerequisites¶

Presently, NSC has been tested and shown to work on these platforms: Linux i386 and Linux x86-64. NSC does not support powerpc. Use on FreeBSD or OS X is unsupported (although it may be able to work).

Building NSC requires the packages flex and bison.

NSC requires use of gcc-4.9 or gcc-5 series, and will not build on newer systems lacking the older compilers.

Configuring and Downloading¶

NSC must either be downloaded separately from its own repository, or downloading when using the bake build system of ns-3.

For ns-3.17 through ns-3.28 releases, when using bake, one obtains NSC implicitly as part of an “allinone” configuration, such as:

$ cd bake
$ python bake.py configure -e ns-allinone-3.27
$ python bake.py download
$ python bake.py build

For ns-3.29 and later versions, including the ‘ns-3-allinone’ development version, one must explicitly add NSC (‘nsc-0.5.3’) to the bake configuration, such as:

$ cd bake
$ python bake.py configure -e ns-allinone-3.29 -e nsc-0.5.3
$ python bake.py download
$ python bake.py build

Instead of a released version, one may use the ns-3 development version by specifying “ns-3-allinone” to the configure step above.

NSC may also be downloaded from its download site using Mercurial:

$ hg clone https://secure.wand.net.nz/mercurial/nsc

Prior to the ns-3.17 release, NSC was included in the allinone tarball and the released version did not need to be separately downloaded.

Building and validating¶

NSC may be built as part of the bake build process; alternatively, one may build NSC by itself using its build system; e.g.:

$ cd nsc-dev
$ python scons.py

Once NSC has been built either manually or through the bake system, change into the ns-3 source directory and try running the following configuration:

$ ./waf configure

If NSC has been previously built and found by waf, then you will see:

Network Simulation Cradle     : enabled

If NSC has not been found, you will see:

Network Simulation Cradle     : not enabled (NSC not found (see option --with-nsc))

In this case, you must pass the relative or absolute path to the NSC libraries with the “–with-nsc” configure option; e.g.

$ ./waf configure --with-nsc=/path/to/my/nsc/directory

For ns-3 releases prior to the ns-3.17 release, using the build.py script in ns-3-allinone directory, NSC will be built by default unless the platform does not support it. To explicitly disable it when building ns-3, type:

$ ./waf configure --enable-examples --enable-tests --disable-nsc

If waf detects NSC, then building ns-3 with NSC is performed the same way with waf as without it. Once ns-3 is built, try running the following test suite:

$ ./test.py -s ns3-tcp-interoperability

If NSC has been successfully built, the following test should show up in the results:

PASS TestSuite ns3-tcp-interoperability

This confirms that NSC is ready to use.

Usage¶

There are a few example files. Try:

$ ./waf --run tcp-nsc-zoo
$ ./waf --run tcp-nsc-lfn

These examples will deposit some .pcap files in your directory, which can be examined by tcpdump or wireshark.

Let’s look at the examples/tcp/tcp-nsc-zoo.cc file for some typical usage. How does it differ from using native ns-3 TCP? There is one main configuration line, when using NSC and the ns-3 helper API, that needs to be set:

InternetStackHelper internetStack;
internetStack.SetNscStack ("liblinux2.6.26.so");
// this switches nodes 0 and 1 to NSCs Linux 2.6.26 stack.
internetStack.Install (n.Get(0));
internetStack.Install (n.Get(1));

The key line is the SetNscStack. This tells the InternetStack helper to aggregate instances of NSC TCP instead of native ns-3 TCP to the remaining nodes. It is important that this function be called before calling the Install() function, as shown above.

Which stacks are available to use? Presently, the focus has been on Linux 2.6.18 and Linux 2.6.26 stacks for ns-3. To see which stacks were built, one can execute the following find command at the ns-3 top level directory:

$ find nsc -name "*.so" -type f
nsc/linux-2.6.18/liblinux2.6.18.so
nsc/linux-2.6.26/liblinux2.6.26.so

This tells us that we may either pass the library name liblinux2.6.18.so or liblinux2.6.26.so to the above configuration step.

Stack configuration¶

NSC TCP shares the same configuration attributes that are common across TCP sockets, as described above and documented in Doxygen

Additionally, NSC TCP exports a lot of configuration variables into the ns-3 attributes system, via a sysctl-like interface. In the examples/tcp/tcp-nsc-zoo example, you can see the following configuration:

// this disables TCP SACK, wscale and timestamps on node 1 (the attributes


  represent sysctl-values).
Config::Set ("/NodeList/1/$ns3::Ns3NscStack<linux2.6.26>/net.ipv4.tcp_sack",


  StringValue ("0"));
Config::Set ("/NodeList/1/$ns3::Ns3NscStack<linux2.6.26>/net.ipv4.tcp_timestamps",
StringValue ("0"));
Config::Set ("/NodeList/1/$ns3::Ns3NscStack<linux2.6.26>/net.ipv4.tcp_window_scaling",
StringValue ("0"));

These additional configuration variables are not available to native ns-3 TCP.

Also note that default values for TCP attributes in ns-3 TCP may differ from the NSC TCP implementation. Specifically in ns-3:

Therefore when making comparisons between results obtained using NSC and ns-3 TCP, care must be taken to ensure these values are set appropriately. See /examples/tcp/tcp-nsc-comparison.cc for an example.

NSC API¶

This subsection describes the API that NSC presents to ns-3 or any other simulator. NSC provides its API in the form of a number of classes that are defined in sim/sim_interface.h in the nsc directory.

ns-3 implementation¶

The ns-3 implementation makes use of the above NSC API, and is implemented as follows.

The three main parts are:

src/internet/model/nsc-tcp-l4-protocol is the main class. Upon Initialization, it loads an NSC network stack to use (via dlopen()). Each instance of this class may use a different stack. The stack (=shared library) to use is set using the SetNscLibrary() method (at this time its called indirectly via the internet stack helper). The NSC stack is then set up accordingly (timers etc). The NscTcpL4Protocol::Receive() function hands the packet it receives (must be a complete TCP/IP packet) to the NSC stack for further processing. To be able to send packets, this class implements the NSC send_callback() method. This method is called by NSC whenever the NSC stack wishes to send a packet out to the network. Its arguments are a raw buffer, containing a complete TCP/IP packet, and a length value. This method therefore has to convert the raw data to a Ptr<Packet> usable by ns-3. In order to avoid various IPv4 header issues, the NSC IP header is not included. Instead, the TCP header and the actual payload are put into the Ptr<Packet>, after this the Packet is passed down to layer 3 for sending the packet out (no further special treatment is needed in the send code path).

This class calls ns3::NscTcpSocketImpl both from the NSC wakeup() callback and from the receive path (to ensure that possibly queued data is scheduled for sending).

src/internet/model/nsc-tcp-socket-impl implements the NSC socket interface. Each instance has its own m_nscTcpSocket. Data that is sent will be handed to the NSC stack via m_nscTcpSocket->send_data() (and not to NscTcpL4Protocol, this is the major difference compared to ns-3 TCP). The class also queues up data that is sent before the underlying descriptor has entered an ESTABLISHED state. This class is called from the NscTcpL4Protocol class, when the NscTcpL4Protocol wakeup() callback is invoked by NSC. NscTcpSocketImpl then checks the current connection state (SYN_SENT, ESTABLISHED, LISTEN…) and schedules appropriate callbacks as needed, e.g. a LISTEN socket will schedule accept() to see if a new connection must be accepted, an ESTABLISHED socket schedules any pending data for writing, schedule a read() callback, etc.

Note that ns3::NscTcpSocketImpl does not interact with NSC TCP directly: instead, data is redirected to NSC. NSC TCP calls the NSC TCP sockets of a node when its wakeup() callback is invoked by NSC.

Limitations¶

For more information, see this wiki page.

UDP model in ns-3¶

This chapter describes the UDP model available in ns-3.

Generic support for UDP¶

ns-3 supports a native implementation of UDP. It provides a connectionless, unreliable datagram packet service. Packets may be reordered or duplicated before they arrive. UDP calculates and checks checksums to catch transmission errors.

This implementation inherits from a few common header classes in the src/network directory, so that user code can swap out implementations with minimal changes to the scripts.

Here are the important abstract base classes:

ns-3 UDP¶

This is an implementation of the User Datagram Protocol described in RFC 768. UDP uses a simple connectionless communication model with a minimum of protocol mechanism. The implementation provides checksums for data integrity, and port numbers for addressing different functions at the source and destination of the datagram. It has no handshaking dialogues, and thus exposes the user’s data to any unreliability of the underlying network. There is no guarantee of data delivery, ordering, or duplicate protection.

Usage¶

In many cases, usage of UDP is set at the application layer by telling the ns-3 application which kind of socket factory to use.

Using the helper functions defined in src/applications/helper, here is how one would create a UDP receiver:

// Create a packet sink on the receiver
uint16_t port = 50000;
Address sinkLocalAddress(InetSocketAddress (Ipv4Address::GetAny (), port));
PacketSinkHelper sinkHelper ("ns3::UdpSocketFactory", sinkLocalAddress);
ApplicationContainer sinkApp = sinkHelper.Install (serverNode);
sinkApp.Start (Seconds (1.0));
sinkApp.Stop (Seconds (10.0));

Similarly, the below snippet configures OnOffApplication traffic source to use UDP:

// Create the OnOff applications to send data to the UDP receiver
OnOffHelper clientHelper ("ns3::UdpSocketFactory", Address ());
clientHelper.SetAttribute ("Remote", remoteAddress);
ApplicationContainer clientApps = (clientHelper.Install (clientNode);
clientApps.Start (Seconds (2.0));
clientApps.Stop (Seconds (9.0));

For users who wish to have a pointer to the actual socket (so that socket operations like Bind(), setting socket options, etc. can be done on a per-socket basis), UDP sockets can be created by using the Socket::CreateSocket() method as given below:

Ptr<Node> node = CreateObject<Node> ();
InternetStackHelper internet;
internet.Install (node);
Ptr<SocketFactory> socketFactory = node->GetObject<UdpSocketFactory> ();
Ptr<Socket> socket = socketFactory->CreateSocket ();
socket->Bind (InetSocketAddress (Ipv4Address::GetAny (), 80));

Once a UDP socket is created, we do not need an explicit connection setup before sending and receiving data. Being a connectionless protocol, all we need to do is to create a socket and bind it to a known port. For a client, simply create a socket and start sending data. The Bind() call allows an application to specify a port number and an address on the local machine. It allocates a local IPv4 endpoint for this socket.

At the end of data transmission, the socket is closed using the Socket::Close(). It returns a 0 on success and -1 on failure.

Please note that applications usually create the sockets automatically. Please refer to the source code of your preferred application to discover how and when it creates the socket.

UDP Socket interaction and interface with Application layer¶

The following is the description of the public interface of the UDP socket, and how the interface is used to interact with the socket itself.

Socket APIs for UDP connections:

----

Public callbacks

These callbacks are called by the UDP socket to notify the application of interesting events. We will refer to these with the protected name used in socket.h, but we will provide the API function to set the pointers to these callback as well.

Validation¶

The following test cases have been provided for UDP implementation in the src/internet/test/udp-test.cc file.

Limitations¶

Internet Applications Module Documentation¶

The goal of this module is to hold all the Internet-specific applications, and most notably some very specific applications (e.g., ping) or daemons (e.g., radvd). Other non-Internet-specific applications such as packet generators are contained in other modules.

The source code for the new module lives in the directory src/internet-apps.

Each application has its own goals, limitations and scope, which are briefly explained in the following.

All the applications are extensively used in the top-level examples directories. The users are encouraged to check the scripts therein to have a clear overview of the various options and usage tricks.

V4Ping¶

This app mimics a “ping” (ICMP Echo) using IPv4. The application allows the following attributes to be set:

Moreover, the user can access the measured RTT value (as a Traced Source).

Ping6¶

This app mimics a “ping” (ICMP Echo) using IPv6. The application allows the following attributes to be set:

Radvd¶

This app mimics a “RADVD” daemon. I.e., the daemon responsible for IPv6 routers advertisements. All the IPv6 routers should have a RADVD daemon installed.

The configuration of the Radvd application mimics the one of the radvd Linux program.

DHCPv4¶

The ns-3 implementation of Dynamic Host Configuration Protocol (DHCP) follows the specifications of RFC 2131 and RFC 2132.

The source code for DHCP is located in src/internet-apps/model and consists of the following 6 files:

Helpers¶

The following two files have been added to src/internet-apps/helper for DHCP:

Tests¶

The tests for DHCP can be found at src/internet-apps/test/dhcp-test.cc

Examples¶

The examples for DHCP can be found at src/internet-apps/examples/dhcp-example.cc

Scope and Limitations¶

The server should be provided with a network address, mask and a range of address for the pool. One client application can be installed on only one netdevice in a node, and can configure address for only that netdevice.

The following five basic DHCP messages are supported:

Also, the following eight options of BootP are supported:

The client identifier option (61) can be implemented in near future.

In the current implementation, a DHCP client can obtain IPv4 address dynamically from the DHCP server, and can renew it within a lease time period.

Multiple DHCP servers can be configured, but the implementation does not support the use of a DHCP Relay yet. PageBreak

Model Description¶

The source code for the lr-wpan module lives in the directory src/lr-wpan.

Design¶

The model design closely follows the standard from an architectural standpoint.

APIs¶

The APIs closely follow the standard, adapted for ns-3 naming conventions and idioms. The APIs are organized around the concept of service primitives as shown in the following figure adapted from Figure 14 of IEEE Std. 802.15.4-2006.

MCPS-DATA.request      (


                        SrcAddrMode,


                        DstAddrMode,


                        DstPANId,


                        DstAddr,


                        msduLength,


                        msdu,


                        msduHandle,


                        TxOptions,


                        SecurityLevel,


                        KeyIdMode,


                        KeySource,


                        KeyIndex


                        )

struct McpsDataRequestParameters
{


  uint8_t m_srcAddrMode;


  uint8_t m_dstAddrMode;


  ...
};
void
LrWpanMac::McpsDataRequest (McpsDataRequestParameters params)
{
...
}

MAC¶

The MAC at present implements the unslotted CSMA/CA variant, without beaconing. Currently there is no support for coordinators and the relevant APIs.

The implemented MAC is similar to Contiki’s NullMAC, i.e., a MAC without sleep features. The radio is assumed to be always active (receiving or transmitting), of completely shut down. Frame reception is not disabled while performing the CCA.

The main API supported is the data transfer API (McpsDataRequest/Indication/Confirm). CSMA/CA according to Stc 802.15.4-2006, section 7.5.1.4 is supported. Frame reception and rejection according to Std 802.15.4-2006, section 7.5.6.2 is supported, including acknowledgements. Only short addressing completely implemented. Various trace sources are supported, and trace sources can be hooked to sinks.

PHY¶

The physical layer components consist of a Phy model, an error rate model, and a loss model. The error rate model presently models the error rate for IEEE 802.15.4 2.4 GHz AWGN channel for OQPSK; the model description can be found in IEEE Std 802.15.4-2006, section E.4.1.7. The Phy model is based on SpectrumPhy and it follows specification described in section 6 of IEEE Std 802.15.4-2006. It models PHY service specifications, PPDU formats, PHY constants and PIB attributes. It currently only supports the transmit power spectral density mask specified in 2.4 GHz per section 6.5.3.1. The noise power density assumes uniformly distributed thermal noise across the frequency bands. The loss model can fully utilize all existing simple (non-spectrum phy) loss models. The Phy model uses the existing single spectrum channel model. The physical layer is modeled on packet level, that is, no preamble/SFD detection is done. Packet reception will be started with the first bit of the preamble (which is not modeled), if the SNR is more than -5 dB, see IEEE Std 802.15.4-2006, appendix E, Figure E.2. Reception of the packet will finish after the packet was completely transmitted. Other packets arriving during reception will add up to the interference/noise.

Currently the receiver sensitivity is set to a fixed value of -106.58 dBm. This corresponds to a packet error rate of 1% for 20 byte reference packets for this signal power, according to IEEE Std 802.15.4-2006, section 6.1.7. In the future we will provide support for changing the sensitivity to different values.

NetDevice¶

Although it is expected that other technology profiles (such as 6LoWPAN and ZigBee) will write their own NetDevice classes, a basic LrWpanNetDevice is provided, which encapsulates the common operations of creating a generic LrWpan device and hooking things together.

MAC addresses¶

Contrary to other technologies, a IEEE 802.15.4 has 2 different kind of addresses:

The 64-bit addresses are unique worldwide, and set by the device vendor (in a real device). The 16-bit addresses are not guaranteed to be unique, and they are typically either assigned during the devices deployment, or assigned dynamically during the device bootstrap.

In ns-3 the device bootstrap is not (yet) present. Hence, both addresses are set when the device is created.

The other relavant “address” to consider is the PanId (16 bits), which represents the PAN the device is attached to.

Due to the limited number of available bytes in a packet, IEEE 802.15.4 tries to use short addresses instead of long addresses, even though the two might be used at the same time.

For the sake of communicating with the upper layers, and in particular to generate auto-configured IPv6 addresses, each NetDevice must identify itself with a MAC address. The MAC addresses are also used during packet reception, so it is important to use them consistently.

Focusing on IPv6 Stateless address autoconfiguration (SLAAC), there are two relevant RFCs to consider: RFC 4944 and RFC 6282, and the two differ on how to build the IPv6 address given the NetDevice address.

RFC 4944 mandates that the IID part of the IPv6 address is calculated as YYYY:00ff:fe00:XXXX, while RFC 6282 mandates that the IID part of the IPv6 address is calculated as 0000:00ff:fe00:XXXX where XXXX is the device short address, and YYYY is the PanId. In both cases the U/L bit must be set to local, so in the RFC 4944 the PanId might have one bit flipped.

In order to facilitate interoperability, and to avoid unwanted module dependencies, the ns-3 implementation moves the IID calculation in the LrWpanNetDevice::GetAddress (), which will return an Address formatted properly, i.e.:

The 48-bit pseudo-address is generated according to either RFC 4944 or RFC 6282 depending on the configuration of an Attribute (PseudoMacAddressMode).

The default is to use RFC 6282 style addresses.

Note that, on reception, a packet might contain either a short or a long address. This is reflected in the upper-layer notification callback, which can contain either the pseudo-address (48 bits) or the long address (64 bit) of the sender.

Note also that RFC 4944 or RFC 6282 are the RFCs defining the IPv6 address compression formats (HC1 and IPHC respectively). It is defintely not a good idea to either mix devices using different pseudo-address format or compression types in the same network. This point is further discussed in the sixlowpan module documentation.

Scope and Limitations¶

Future versions of this document will contain a PICS proforma similar to Appendix D of IEEE 802.15.4-2006. The current emphasis is on the unslotted mode of 802.15.4 operation for use in Zigbee, and the scope is limited to enabling a single mode (CSMA/CA) with basic data transfer capabilities. Association with PAN coordinators is not yet supported, nor the use of extended addressing. Interference is modeled as AWGN but this is currently not thoroughly tested.

The NetDevice Tx queue is not limited, i.e., packets are never dropped due to queue becoming full. They may be dropped due to excessive transmission retries or channel access failure.

References¶

Usage¶

Enabling lr-wpan¶

Add lr-wpan to the list of modules built with ns-3.

Helper¶

The helper is patterned after other device helpers. In particular, tracing (ascii and pcap) is enabled similarly, and enabling of all lr-wpan log components is performed similarly. Use of the helper is exemplified in examples/lr-wpan-data.cc. For ascii tracing, the transmit and receive traces are hooked at the Mac layer.

The default propagation loss model added to the channel, when this helper is used, is the LogDistancePropagationLossModel with default parameters.

Examples¶

The following examples have been written, which can be found in src/lr-wpan/examples/:

In particular, the module enables a very simplified end-to-end data transfer scenario, implemented in lr-wpan-data.cc. The figure shows a sequence of events that are triggered when the MAC receives a DataRequest from the higher layer. It invokes a Clear Channel Assessment (CCA) from the PHY, and if successful, sends the frame down to the PHY where it is transmitted over the channel and results in a DataIndication on the peer node.

Tests¶

The following tests have been written, which can be found in src/lr-wpan/tests/:

Validation¶

The model has not been validated against real hardware. The error model has been validated against the data in IEEE Std 802.15.4-2006, section E.4.1.7 (Figure E.2). The MAC behavior (CSMA backoff) has been validated by hand against expected behavior. The below plot is an example of the error model validation and can be reproduced by running lr-wpan-error-model-plot.cc:

Design Documentation¶

Overview¶

An overview of the LTE-EPC simulation model is depicted in the figure Overview of the LTE-EPC simulation model. There are two main components:

Design Criteria¶

LTE Model¶

The LTE model has been designed to support the evaluation of the following aspects of LTE systems:

In order to model LTE systems to a level of detail that is sufficient to allow a correct evaluation of the above mentioned aspects, the following requirements have been considered:

EPC Model¶

The main objective of the EPC model is to provides means for the simulation of end-to-end IP connectivity over the LTE model. To this aim, it supports for the interconnection of multiple UEs to the Internet, via a radio access network of multiple eNBs connected to the core network, as shown in Figure Overview of the LTE-EPC simulation model.

The following design choices have been made for the EPC model:

Architecture¶

LTE Model¶

UE architecture¶

The architecture of the LTE radio protocol stack model of the UE is represented in the figures LTE radio protocol stack architecture for the UE on the data plane and LTE radio protocol stack architecture for the UE on the control plane which highlight respectively the data plane and the control plane.

eNB architecture¶

The architecture of the LTE radio protocol stack model of the eNB is represented in the figures LTE radio protocol stack architecture for the eNB on the data plane and LTE radio protocol stack architecture for the eNB on the control plane which highlight respectively the data plane and the control plane.

EPC Model¶

EPC data plane¶

In Figure LTE-EPC data plane protocol stack, we represent the end-to-end LTE-EPC data plane protocol stack as it is modeled in the simulator. The figure shows all nodes in the data path, i.e. UE, eNB, SGW, PGW and a remote host in the Internet. All protocol stacks (S5 protocol stack, S1-U protocol stack and the LTE radio protocol stack) specified by 3GPP are present.

EPC control plane¶

The architecture of the implementation of the control plane model is shown in figure LTE-EPC control plane protocol stack. The control interfaces that are modeled explicitly are the S1-MME, the S11, and the S5 interfaces. The X2 interface is also modeled explicitly and it is described in more detail in section X2

The S1-MME, the S11 and the S5 interfaces are modeled using procotol data units sent over its respective links. These interfaces use the SCTP protocol as transport protocol but currently, the SCTP protocol is not modeled in the ns-3 simulator, so the UDP protocol is used instead of the SCTP protocol.

Channel and Propagation¶

For channel modeling purposes, the LTE module uses the SpectrumChannel interface provided by the spectrum module. At the time of this writing, two implementations of such interface are available: SingleModelSpectrumChannel and MultiModelSpectrumChannel, and the LTE module requires the use of the MultiModelSpectrumChannel in order to work properly. This is because of the need to support different frequency and bandwidth configurations. All the propagation models supported by MultiModelSpectrumChannel can be used within the LTE module.

Use of the Buildings model with LTE¶

The recommended propagation model to be used with the LTE module is the one provided by the Buildings module, which was in fact designed specifically with LTE (though it can be used with other wireless technologies as well). Please refer to the documentation of the Buildings module for generic information on the propagation model it provides.

In this section we will highlight some considerations that specifically apply when the Buildings module is used together with the LTE module.

The naming convention used in the following will be:

The LTE module considers FDD only, and implements downlink and uplink propagation separately. As a consequence, the following pathloss computations are performed

The LTE model does not provide the following pathloss computations:

The Buildings model does not know the actual type of the node; i.e., it is not aware of whether a transmitter node is a UE, a MBS, or a SC. Rather, the Buildings model only cares about the position of the node: whether it is indoor and outdoor, and what is its z-axis respect to the rooftop level. As a consequence, for an eNB node that is placed outdoor and at a z-coordinate above the rooftop level, the propagation models typical of MBS will be used by the Buildings module. Conversely, for an eNB that is placed outdoor but below the rooftop, or indoor, the propagation models typical of pico and femtocells will be used.

For communications involving at least one indoor node, the corresponding wall penetration losses will be calculated by the Buildings model. This covers the following use cases:

Please refer to the documentation of the Buildings module for details on the actual models used in each case.

Fading Model¶

The LTE module includes a trace-based fading model derived from the one developed during the GSoC 2010 [Piro2011]. The main characteristic of this model is the fact that the fading evaluation during simulation run-time is based on per-calculated traces. This is done to limit the computational complexity of the simulator. On the other hand, it needs huge structures for storing the traces; therefore, a trade-off between the number of possible parameters and the memory occupancy has to be found. The most important ones are:

With respect to the mathematical channel propagation model, we suggest the one provided by the rayleighchan function of Matlab, since it provides a well accepted channel modelization both in time and frequency domain. For more information, the reader is referred to [mathworks].

The simulator provides a matlab script (src/lte/model/fading-traces/fading-trace-generator.m) for generating traces based on the format used by the simulator. In detail, the channel object created with the rayleighchan function is used for filtering a discrete-time impulse signal in order to obtain the channel impulse response. The filtering is repeated for different TTI, thus yielding subsequent time-correlated channel responses (one per TTI). The channel response is then processed with the pwelch function for obtaining its power spectral density values, which are then saved in a file with the proper format compatible with the simulator model.

Since the number of variable it is pretty high, generate traces considering all of them might produce a high number of traces of huge size. On this matter, we considered the following assumptions of the parameters based on the 3GPP fading propagation conditions (see Annex B.2 of [TS36104]):

According to the parameters we considered, the following formula express in detail the total size S_{traces} of the fading traces:

S_{traces} = S_{sample} \times N_{RB} \times \frac{T_{trace}}{T_{sample}} \times N_{scenarios} \mbox{ [bytes]}

where S_{sample} is the size in bytes of the sample (e.g., 8 in case of double precision, 4 in case of float precision), N_{RB} is the number of RB or set of RBs to be considered, T_{trace} is the total length of the trace, T_{sample} is the time resolution of the trace (1 ms), and N_{scenarios} is the number of fading scenarios that are desired (i.e., combinations of different sets of channel taps and user speed values). We provide traces for 3 different scenarios one for each taps configuration defined in Annex B.2 of [TS36104]:

hence N_{scenarios} = 3. All traces have T_{trace} = 10 s and RB_{NUM} = 100. This results in a total 24 MB bytes of traces.

Antennas¶

Being based on the SpectrumPhy, the LTE PHY model supports antenna modeling via the ns-3 AntennaModel class. Hence, any model based on this class can be associated with any eNB or UE instance. For instance, the use of the CosineAntennaModel associated with an eNB device allows to model one sector of a macro base station. By default, the IsotropicAntennaModel is used for both eNBs and UEs.

PHY¶

Overview¶

The physical layer model provided in this LTE simulator is based on the one described in [Piro2011], with the following modifications. The model now includes the inter cell interference calculation and the simulation of uplink traffic, including both packet transmission and CQI generation.

Subframe Structure¶

The subframe is divided into control and data part as described in Figure LTE subframe division..

MAC to Channel delay¶

To model the latency of real MAC and PHY implementations, the PHY model simulates a MAC-to-channel delay in multiples of TTIs (1ms). The transmission of both data and control packets are delayed by this amount.

CQI feedback¶

The generation of CQI feedback is done accordingly to what specified in [FFAPI]. In detail, we considered the generation of periodic wideband CQI (i.e., a single value of channel state that is deemed representative of all RBs in use) and inband CQIs (i.e., a set of value representing the channel state for each RB).

The CQI index to be reported is obtained by first obtaining a SINR measurement and then passing this SINR measurement to the Adaptive Modulation and Coding module which will map it to the CQI index.

In downlink, the SINR used to generate CQI feedback can be calculated in two different ways:

To switch between this two CQI generation approaches, LteHelper::UsePdschForCqiGeneration needs to be configured: false for first approach and true for second approach (true is default value):

Config::SetDefault ("ns3::LteHelper::UsePdschForCqiGeneration", BooleanValue (true));

In uplink, two types of CQIs are implemented:

The scheduler interface include an attribute system called UlCqiFilter for managing the filtering of the CQIs according to their nature, in detail:

It has to be noted that, the FfMacScheduler provides only the interface and it is matter of the actual scheduler implementation to include the code for managing these attributes (see scheduler related section for more information on this matter).

Interference Model¶

The PHY model is based on the well-known Gaussian interference models, according to which the powers of interfering signals (in linear units) are summed up together to determine the overall interference power.

The sequence diagram of Figure Sequence diagram of the PHY interference calculation procedure shows how interfering signals are processed to calculate the SINR, and how SINR is then used for the generation of CQI feedback.

LTE Spectrum Model¶

The usage of the radio spectrum by eNBs and UEs in LTE is described in [TS36101]. In the simulator, radio spectrum usage is modeled as follows. Let f_c denote the LTE Absolute Radio Frequency Channel Number, which identifies the carrier frequency on a 100 kHz raster; furthermore, let B be the Transmission Bandwidth Configuration in number of Resource Blocks. For every pair (f_c,B) used in the simulation we define a corresponding SpectrumModel using the functionality provided by the Spectrum Module . model using the Spectrum framework described in [Baldo2009]. f_c and B can be configured for every eNB instantiated in the simulation; hence, each eNB can use a different spectrum model. Every UE will automatically use the spectrum model of the eNB it is attached to. Using the MultiModelSpectrumChannel described in [Baldo2009], the interference among eNBs that use different spectrum models is properly accounted for. This allows to simulate dynamic spectrum access policies, such as for example the spectrum licensing policies that are discussed in [Ofcom2600MHz].

Data PHY Error Model¶

The simulator includes an error model of the data plane (i.e., PDSCH and PUSCH) according to the standard link-to-system mapping (LSM) techniques. The choice is aligned with the standard system simulation methodology of OFDMA radio transmission technology. Thanks to LSM we are able to maintain a good level of accuracy and at the same time limiting the computational complexity increase. It is based on the mapping of single link layer performance obtained by means of link level simulators to system (in our case network) simulators. In particular link the layer simulator is used for generating the performance of a single link from a PHY layer perspective, usually in terms of code block error rate (BLER), under specific static conditions. LSM allows the usage of these parameters in more complex scenarios, typical of system/network simulators, where we have more links, interference and “colored” channel propagation phenomena (e.g., frequency selective fading).

To do this the Vienna LTE Simulator [ViennaLteSim] has been used for what concerns the extraction of link layer performance and the Mutual Information Based Effective SINR (MIESM) as LSM mapping function using part of the work recently published by the Signet Group of University of Padua [PaduaPEM].

MIESM¶

The specific LSM method adopted is the one based on the usage of a mutual information metric, commonly referred to as the mutual information per per coded bit (MIB or MMIB when a mean of multiples MIBs is involved). Another option would be represented by the Exponential ESM (EESM); however, recent studies demonstrate that MIESM outperforms EESM in terms of accuracy [LozanoCost].

BLER Curves¶

On this respect, we reused part of the curves obtained within [PaduaPEM]. In detail, we introduced the CB size dependency to the CB BLER curves with the support of the developers of [PaduaPEM] and of the LTE Vienna Simulator. In fact, the module released provides the link layer performance only for what concerns the MCSs (i.e, with a given fixed ECR). In detail the new error rate curves for each has been evaluated with a simulation campaign with the link layer simulator for a single link with AWGN noise and for CB size of 104, 140, 256, 512, 1024, 2048, 4032 and 6144. These curves has been mapped with the Gaussian cumulative model formula presented above for obtaining the correspondents b_{ECR} and c_{ECR} parameters.

The BLER performance of all MCS obtained with the link level simulator are plotted in the following figures (blue lines) together with their correspondent mapping to the Gaussian cumulative distribution (red dashed lines).

Integration of the BLER curves in the ns-3 LTE module¶

The model implemented uses the curves for the LSM of the recently LTE PHY Error Model released in the ns3 community by the Signet Group [PaduaPEM] and the new ones generated for different CB sizes. The LteSpectrumPhy class is in charge of evaluating the TB BLER thanks to the methods provided by the LteMiErrorModel class, which is in charge of evaluating the TB BLER according to the vector of the perceived SINR per RB, the MCS and the size in order to proper model the segmentation of the TB in CBs. In order to obtain the vector of the perceived SINRs for data and control signals, two instances of LteChunkProcessor (dedicated to evaluate the SINR for obtaining physical error performance) have been attached to UE downlink and eNB uplink LteSpectrumPhy modules for evaluating the error model distribution of PDSCH (UE side) and ULSCH (eNB side).

The model can be disabled for working with a zero-losses channel by setting the DataErrorModelEnabled attribute of the LteSpectrumPhy class (by default is active). This can be done according to the standard ns3 attribute system procedure, that is:

Config::SetDefault ("ns3::LteSpectrumPhy::DataErrorModelEnabled", BooleanValue (false));

Control Channels PHY Error Model¶

The simulator includes the error model for downlink control channels (PCFICH and PDCCH), while in uplink it is assumed and ideal error-free channel. The model is based on the MIESM approach presented before for considering the effects of the frequency selective channel since most of the control channels span the whole available bandwidth.

PCFICH + PDCCH Error Model¶

The model adopted for the error distribution of these channels is based on an evaluation study carried out in the RAN4 of 3GPP, where different vendors investigated the demodulation performance of the PCFICH jointly with PDCCH. This is due to the fact that the PCFICH is the channel in charge of communicating to the UEs the actual dimension of the PDCCH (which spans between 1 and 3 symbols); therefore the correct decodification of the DCIs depends on the correct interpretation of both ones. In 3GPP this problem have been evaluated for improving the cell-edge performance [FujitsuWhitePaper], where the interference among neighboring cells can be relatively high due to signal degradation. A similar problem has been notices in femto-cell scenario and, more in general, in HetNet scenarios the bottleneck has been detected mainly as the PCFICH channel [Bharucha2011], where in case of many eNBs are deployed in the same service area, this channel may collide in frequency, making impossible the correct detection of the PDCCH channel, too.

In the simulator, the SINR perceived during the reception has been estimated according to the MIESM model presented above in order to evaluate the error distribution of PCFICH and PDCCH. In detail, the SINR samples of all the RBs are included in the evaluation of the MI associated to the control frame and, according to this values, the effective SINR (eSINR) is obtained by inverting the MI evaluation process. It has to be noted that, in case of MIMO transmission, both PCFICH and the PDCCH use always the transmit diversity mode as defined by the standard. According to the eSINR perceived the decodification error probability can be estimated as function of the results presented in [R4-081920]. In case an error occur, the DCIs discarded and therefore the UE will be not able to receive the correspondent Tbs, therefore resulting lost.

MIMO Model¶

The use of multiple antennas both at transmitter and receiver side, known as multiple-input and multiple-output (MIMO), is a problem well studied in literature during the past years. Most of the work concentrate on evaluating analytically the gain that the different MIMO schemes might have in term of capacity; however someones provide also information of the gain in terms of received power [CatreuxMIMO].

According to the considerations above, a model more flexible can be obtained considering the gain that MIMO schemes bring in the system from a statistical point of view. As highlighted before, [CatreuxMIMO] presents the statistical gain of several MIMO solutions respect to the SISO one in case of no correlation between the antennas. In the work the gain is presented as the cumulative distribution function (CDF) of the output SINR for what concern SISO, MIMO-Alamouti, MIMO-MMSE, MIMO-OSIC-MMSE and MIMO-ZF schemes. Elaborating the results, the output SINR distribution can be approximated with a log-normal one with different mean and variance as function of the scheme considered. However, the variances are not so different and they are approximatively equal to the one of the SISO mode already included in the shadowing component of the BuildingsPropagationLossModel, in detail:

Therefore the PHY layer implements the MIMO model as the gain perceived by the receiver when using a MIMO scheme respect to the one obtained using SISO one. We note that, these gains referred to a case where there is no correlation between the antennas in MIMO scheme; therefore do not model degradation due to paths correlation.

UE PHY Measurements Model¶

According to [TS36214], the UE has to report a set of measurements of the eNBs that the device is able to perceive: the reference signal received power (RSRP) and the reference signal received quality (RSRQ). The former is a measure of the received power of a specific eNB, while the latter includes also channel interference and thermal noise. The UE has to report the measurements jointly with the physical cell identity (PCI) of the cell. Both the RSRP and RSRQ measurements are performed during the reception of the RS, while the PCI is obtained with the Primary Synchronization Signal (PSS). The PSS is sent by the eNB each 5 subframes and in detail in the subframes 1 and 6. In real systems, only 504 distinct PCIs are available, and hence it could occur that two nearby eNBs use the same PCI; however, in the simulator we model PCIs using simulation metadata, and we allow up to 65535 distinct PCIs, thereby avoiding PCI collisions provided that less that 65535 eNBs are simulated in the same scenario.

According to [TS36133] sections 9.1.4 and 9.1.7, RSRP is reported by PHY layer in dBm while RSRQ in dB. The values of RSRP and RSRQ are provided to higher layers through the C-PHY SAP (by means of UeMeasurementsParameters struct) every 200 ms as defined in [TS36331]. Layer 1 filtering is performed by averaging the all the measurements collected during the last window slot. The periodicity of reporting can be adjusted for research purposes by means of the LteUePhy::UeMeasurementsFilterPeriod attribute.

The formulas of the RSRP and RSRQ can be simplified considering the assumption of the PHY layer that the channel is flat within the RB, the finest level of accuracy. In fact, this implies that all the REs within a RB have the same power, therefore:

RSRP = \frac{\sum_{k=0}^{K-1}\frac{\sum_{m=0}^{M-1}(P(k,m))}{M}}{K}

= \frac{\sum_{k=0}^{K-1}\frac{(M \times P(k))}{M}}{K}
= \frac{\sum_{k=0}^{K-1}(P(k))}{K}

where P(k,m) represents the signal power of the RE m within the RB k, which, as observed before, is constant within the same RB and equal to P(k), M is the number of REs carrying the RS in a RB and K is the number of RBs. It is to be noted that P(k), and in general all the powers defined in this section, is obtained in the simulator from the PSD of the RB (which is provided by the LteInterferencePowerChunkProcessor), in detail:

P(k) = PSD_{RB}(k)*180000/12

where PSD_{RB}(k) is the power spectral density of the RB k, 180000 is the bandwidth in Hz of the RB and 12 is the number of REs per RB in an OFDM symbol. Similarly, for RSSI we have

RSSI = \sum_{k=0}^{K-1} \frac{\sum_{s=0}^{S-1} \sum_{r=0}^{R-1}( P(k,s,r) + I(k,s,r) + N(k,s,r))}{S}

where S is the number of OFDM symbols carrying RS in a RB and R is the number of REs carrying a RS in a OFDM symbol (which is fixed to 2) while P(k,s,r), I(k,s,r) and N(k,s,r) represent respectively the perceived power of the serving cell, the interference power and the noise power of the RE r in symbol s. As for RSRP, the measurements within a RB are always equals among each others according to the PHY model; therefore P(k,s,r) = P(k), I(k,s,r) = I(k) and N(k,s,r) = N(k), which implies that the RSSI can be calculated as:

RSSI = \sum_{k=0}^{K-1} \frac{S \times 2 \times ( P(k) + I(k) + N(k))}{S}

= \sum_{k=0}^{K-1} 2 \times ( P(k) + I(k) + N (k))

Considering the constraints of the PHY reception chain implementation, and in order to maintain the level of computational complexity low, only RSRP can be directly obtained for all the cells. This is due to the fact that LteSpectrumPhy is designed for evaluating the interference only respect to the signal of the serving eNB. This implies that the PHY layer is optimized for managing the power signals information with the serving eNB as a reference. However, RSRP and RSRQ of neighbor cell i can be extracted by the current information available of the serving cell j as detailed in the following:

RSRP_i = \frac{\sum_{k=0}^{K-1}(P_i(k))}{K}

RSSI_i = RSSI_j = \sum_{k=0}^{K-1} 2 \times ( I_j(k) + P_j(k) + N_j(k) )

RSRQ_i^j = K \times RSRP_i / RSSI_j

where RSRP_i is the RSRP of the neighbor cell i, P_i(k) is the power perceived at any RE within the RB k, K is the total number of RBs, RSSI_i is the RSSI of the neighbor cell i when the UE is attached to cell j (which, since it is the sum of all the received powers, coincides with RSSI_j), I_j(k) is the total interference perceived by UE in any RE of RB k when attached to cell i (obtained by the LteInterferencePowerChunkProcessor), P_j(k) is the power perceived of cell j in any RE of the RB k and N is the power noise spectral density in any RE. The sample is considered as valid in case of the RSRQ evaluated is above the LteUePhy::RsrqUeMeasThreshold attribute.

HARQ¶

The HARQ scheme implemented is based on a incremental redundancy (IR) solutions combined with multiple stop-and-wait processes for enabling a continuous data flow. In detail, the solution adopted is the soft combining hybrid IR Full incremental redundancy (also called IR Type II), which implies that the retransmissions contain only new information respect to the previous ones. The resource allocation algorithm of the HARQ has been implemented within the respective scheduler classes (i.e., RrFfMacScheduler and PfFfMacScheduler, refer to their correspondent sections for more info), while the decodification part of the HARQ has been implemented in the LteSpectrumPhy and LteHarqPhy classes which will be detailed in this section.

According to the standard, the UL retransmissions are synchronous and therefore are allocated 7 ms after the original transmission. On the other hand, for the DL, they are asynchronous and therefore can be allocated in a more flexible way starting from 7 ms and it is a matter of the specific scheduler implementation. The HARQ processes behavior is depicted in Figure:ref:fig-harq-processes-scheme.

At the MAC layer, the HARQ entity residing in the scheduler is in charge of controlling the 8 HARQ processes for generating new packets and managing the retransmissions both for the DL and the UL. The scheduler collects the HARQ feedback from eNB and UE PHY layers (respectively for UL and DL connection) by means of the FF API primitives SchedUlTriggerReq and SchedUlTriggerReq. According to the HARQ feedback and the RLC buffers status, the scheduler generates a set of DCIs including both retransmissions of HARQ blocks received erroneous and new transmissions, in general, giving priority to the former. On this matter, the scheduler has to take into consideration one constraint when allocating the resource for HARQ retransmissions, it must use the same modulation order of the first transmission attempt (i.e., QPSK for MCS \in [0..9], 16QAM for MCS \in [10..16] and 64QAM for MCS \in [17..28]). This restriction comes from the specification of the rate matcher in the 3GPP standard [ TS36212]_, where the algorithm fixes the modulation order for generating the different blocks of the redundancy versions.

The PHY Error Model model (i.e., the LteMiErrorModel class already presented before) has been extended for considering IR HARQ according to [wimaxEmd], where the parameters for the AWGN curves mapping for MIESM mapping in case of retransmissions are given by:

R_{eff} = \frac{X}{\sum\limits_{i=1}^q C_i}

M_{I eff} = \frac{\sum\limits_{i=1}^q C_i M_i}{\sum\limits_{i=1}^q C_i}

where X is the number of original information bits, C_i are number of coded bits, M_i are the mutual information per HARQ block received on the total number of q retransmissions. Therefore, in order to be able to return the error probability with the error model implemented in the simulator evaluates the R_{eff} and the MI_{I eff} and return the value of error probability of the ECR of the same modulation with closest lower rate respect to the R_{eff}. In order to consider the effect of HARQ retransmissions a new sets of curves have been integrated respect to the standard one used for the original MCS. The new curves are intended for covering the cases when the most conservative MCS of a modulation is used which implies the generation of R_{eff} lower respect to the one of standard MCSs. On this matter the curves for 1, 2 and 3 retransmissions have been evaluated for 10 and 17. For MCS 0 we considered only the first retransmission since the produced code rate is already very conservative (i.e., 0.04) and returns an error rate enough robust for the reception (i.e., the downturn of the BLER is centered around -18 dB). It is to be noted that, the size of first TB transmission has been assumed as containing all the information bits to be coded; therefore X is equal to the size of the first TB sent of a an HARQ process. The model assumes that the eventual presence of parity bits in the codewords is already considered in the link level curves. This implies that as soon as the minimum R_{eff} is reached the model is not including the gain due to the transmission of further parity bits.

MAC¶

Resource Allocation Model¶

We now briefly describe how resource allocation is handled in LTE, clarifying how it is modeled in the simulator. The scheduler is in charge of generating specific structures called Data Control Indication (DCI) which are then transmitted by the PHY of the eNB to the connected UEs, in order to inform them of the resource allocation on a per subframe basis. In doing this in the downlink direction, the scheduler has to fill some specific fields of the DCI structure with all the information, such as: the Modulation and Coding Scheme (MCS) to be used, the MAC Transport Block (TB) size, and the allocation bitmap which identifies which RBs will contain the data transmitted by the eNB to each user.

For the mapping of resources to physical RBs, we adopt a localized mapping approach (see [Sesia2009], Section 9.2.2.1); hence in a given subframe each RB is always allocated to the same user in both slots. The allocation bitmap can be coded in different formats; in this implementation, we considered the Allocation Type 0 defined in [TS36213], according to which the RBs are grouped in Resource Block Groups (RBG) of different size determined as a function of the Transmission Bandwidth Configuration in use.

For certain bandwidth values not all the RBs are usable, since the group size is not a common divisor of the group. This is for instance the case when the bandwidth is equal to 25 RBs, which results in a RBG size of 2 RBs, and therefore 1 RB will result not addressable. In uplink the format of the DCIs is different, since only adjacent RBs can be used because of the SC-FDMA modulation. As a consequence, all RBs can be allocated by the eNB regardless of the bandwidth configuration.

Adaptive Modulation and Coding¶

The simulator provides two Adaptive Modulation and Coding (AMC) models: one based on the GSoC model [Piro2011] and one based on the physical error model (described in the following sections).

The former model is a modified version of the model described in [Piro2011], which in turn is inspired from [Seo2004]. Our version is described in the following. Let i denote the generic user, and let \gamma_i be its SINR. We get the spectral efficiency \eta_i of user i using the following equations:

\mathrm{BER} = 0.00005

\Gamma = \frac{ -\ln{ (5 * \mathrm{BER}) } }{ 1.5}

\eta_i = \log_2 { \left( 1 + \frac{ {\gamma}_i }{ \Gamma } \right)}

The procedure described in [R1-081483] is used to get the corresponding MCS scheme. The spectral efficiency is quantized based on the channel quality indicator (CQI), rounding to the lowest value, and is mapped to the corresponding MCS scheme.

Finally, we note that there are some discrepancies between the MCS index in [R1-081483] and that indicated by the standard: [TS36213] Table 7.1.7.1-1 says that the MCS index goes from 0 to 31, and 0 appears to be a valid MCS scheme (TB size is not 0) but in [R1-081483] the first useful MCS index is 1. Hence to get the value as intended by the standard we need to subtract 1 from the index reported in [R1-081483].

The alternative model is based on the physical error model developed for this simulator and explained in the following subsections. This scheme is able to adapt the MCS selection to the actual PHY layer performance according to the specific CQI report. According to their definition, a CQI index is assigned when a single PDSCH TB with the modulation coding scheme and code rate correspondent to that CQI index in table 7.2.3-1 of [TS36213] can be received with an error probability less than 0.1. In case of wideband CQIs, the reference TB includes all the RBGs available in order to have a reference based on the whole available resources; while, for subband CQIs, the reference TB is sized as the RBGs.

Transport Block model¶

The model of the MAC Transport Blocks (TBs) provided by the simulator is simplified with respect to the 3GPP specifications. In particular, a simulator-specific class (PacketBurst) is used to aggregate MAC SDUs in order to achieve the simulator’s equivalent of a TB, without the corresponding implementation complexity. The multiplexing of different logical channels to and from the RLC layer is performed using a dedicated packet tag (LteRadioBearerTag), which performs a functionality which is partially equivalent to that of the MAC headers specified by 3GPP.

The FemtoForum MAC Scheduler Interface¶

This section describes the ns-3 specific version of the LTE MAC Scheduler Interface Specification published by the FemtoForum [FFAPI].

We implemented the ns-3 specific version of the FemtoForum MAC Scheduler Interface [FFAPI] as a set of C++ abstract classes; in particular, each primitive is translated to a C++ method of a given class. The term implemented here is used with the same meaning adopted in [FFAPI], and hence refers to the process of translating the logical interface specification to a particular programming language. The primitives in [FFAPI] are grouped in two groups: the CSCHED primitives, which deal with scheduler configuration, and the SCHED primitives, which deal with the execution of the scheduler. Furthermore, [FFAPI] defines primitives of two different kinds: those of type REQ go from the MAC to the Scheduler, and those of type IND/CNF go from the scheduler to the MAC. To translate these characteristics into C++, we define the following abstract classes that implement Service Access Points (SAPs) to be used to issue the primitives:

There are 3 blocks involved in the MAC Scheduler interface: Control block, Subframe block and Scheduler block. Each of these blocks provide one part of the MAC Scheduler interface. The figure below shows the relationship between the blocks and the SAPs defined in our implementation of the MAC Scheduler Interface.

In addition to the above principles, the following design choices have been taken:

The figure below shows how the MAC Scheduler Interface is used within the eNB.

The User side of both the CSCHED SAP and the SCHED SAP are implemented within the eNB MAC, i.e., in the file lte-enb-mac.cc. The eNB MAC can be used with different scheduler implementations without modifications. The same figure also shows, as an example, how the Round Robin Scheduler is implemented: to interact with the MAC of the eNB, the Round Robin scheduler implements the Provider side of the SCHED SAP and CSCHED SAP interfaces. A similar approach can be used to implement other schedulers as well. A description of each of the scheduler implementations that we provide as part of our LTE simulation module is provided in the following subsections.

Round Robin (RR) Scheduler¶

The Round Robin (RR) scheduler is probably the simplest scheduler found in the literature. It works by dividing the available resources among the active flows, i.e., those logical channels which have a non-empty RLC queue. If the number of RBGs is greater than the number of active flows, all the flows can be allocated in the same subframe. Otherwise, if the number of active flows is greater than the number of RBGs, not all the flows can be scheduled in a given subframe; then, in the next subframe the allocation will start from the last flow that was not allocated. The MCS to be adopted for each user is done according to the received wideband CQIs.

For what concern the HARQ, RR implements the non adaptive version, which implies that in allocating the retransmission attempts RR uses the same allocation configuration of the original block, which means maintaining the same RBGs and MCS. UEs that are allocated for HARQ retransmissions are not considered for the transmission of new data in case they have a transmission opportunity available in the same TTI. Finally, HARQ can be disabled with ns3 attribute system for maintaining backward compatibility with old test cases and code, in detail:

Config::SetDefault ("ns3::RrFfMacScheduler::HarqEnabled", BooleanValue (false));

The scheduler implements the filtering of the uplink CQIs according to their nature with UlCqiFilter attribute, in detail:

Proportional Fair (PF) Scheduler¶

The Proportional Fair (PF) scheduler [Sesia2009] works by scheduling a user when its instantaneous channel quality is high relative to its own average channel condition over time. Let i,j denote generic users; let t be the subframe index, and k be the resource block index; let M_{i,k}(t) be MCS usable by user i on resource block k according to what reported by the AMC model (see Adaptive Modulation and Coding); finally, let S(M, B) be the TB size in bits as defined in [TS36213] for the case where a number B of resource blocks is used. The achievable rate R_{i}(k,t) in bit/s for user i on resource block group k at subframe t is defined as

R_{i}(k,t) = \frac{S\left( M_{i,k}(t), 1\right)}{\tau}

where \tau is the TTI duration. At the start of each subframe t, each RBG is assigned to a certain user. In detail, the index \widehat{i}_{k}(t) to which RBG k is assigned at time t is determined as

\widehat{i}_{k}(t) = \underset{j=1,...,N}{\operatorname{argmax}}

\left( \frac{ R_{j}(k,t) }{ T_\mathrm{j}(t) } \right)

where T_{j}(t) is the past througput performance perceived by the user j. According to the above scheduling algorithm, a user can be allocated to different RBGs, which can be either adjacent or not, depending on the current condition of the channel and the past throughput performance T_{j}(t). The latter is determined at the end of the subframe t using the following exponential moving average approach:

T_{j}(t) =
(1-\frac{1}{\alpha})T_{j}(t-1) +\frac{1}{\alpha} \widehat{T}_{j}(t)

where \alpha is the time constant (in number of subframes) of the exponential moving average, and \widehat{T}_{j}(t) is the actual throughput achieved by the user i in the subframe t. \widehat{T}_{j}(t) is measured according to the following procedure. First we determine the MCS \widehat{M}_j(t) actually used by user j:

\widehat{M}_j(t) = \min_{k: \widehat{i}_{k}(t) = j}{M_{j,k}(t)}

then we determine the total number \widehat{B}_j(t) of RBGs allocated to user j:

\widehat{B}_j(t) = \left| \{ k : \widehat{i}_{k}(t) = j \} \right|

where |\cdot| indicates the cardinality of the set; finally,

\widehat{T}_{j}(t) = \frac{S\left( \widehat{M}_j(t), \widehat{B}_j(t)
\right)}{\tau}

For what concern the HARQ, PF implements the non adaptive version, which implies that in allocating the retransmission attempts the scheduler uses the same allocation configuration of the original block, which means maintaining the same RBGs and MCS. UEs that are allocated for HARQ retransmissions are not considered for the transmission of new data in case they have a transmission opportunity available in the same TTI. Finally, HARQ can be disabled with ns3 attribute system for maintaining backward compatibility with old test cases and code, in detail:

Config::SetDefault ("ns3::PfFfMacScheduler::HarqEnabled", BooleanValue (false));

Maximum Throughput (MT) Scheduler¶

The Maximum Throughput (MT) scheduler [FCapo2012] aims to maximize the overall throughput of eNB. It allocates each RB to the user that can achieve the maximum achievable rate in the current TTI. Currently, MT scheduler in NS-3 has two versions: frequency domain (FDMT) and time domain (TDMT). In FDMT, every TTI, MAC scheduler allocates RBGs to the UE who has highest achievable rate calculated by subband CQI. In TDMT, every TTI, MAC scheduler selects one UE which has highest achievable rate calculated by wideband CQI. Then MAC scheduler allocates all RBGs to this UE in current TTI. The calculation of achievable rate in FDMT and TDMT is as same as the one in PF. Let i,j denote generic users; let t be the subframe index, and k be the resource block index; let M_{i,k}(t) be MCS usable by user i on resource block k according to what reported by the AMC model (see Adaptive Modulation and Coding); finally, let S(M, B) be the TB size in bits as defined in [TS36213] for the case where a number B of resource blocks is used. The achievable rate R_{i}(k,t) in bit/s for user i on resource block k at subframe t is defined as

R_{i}(k,t) = \frac{S\left( M_{i,k}(t), 1\right)}{\tau}

where \tau is the TTI duration. At the start of each subframe t, each RB is assigned to a certain user. In detail, the index \widehat{i}_{k}(t) to which RB k is assigned at time t is determined as

\widehat{i}_{k}(t) = \underset{j=1,...,N}{\operatorname{argmax}}

\left( { R_{j}(k,t) } \right)

When there are several UEs having the same achievable rate, current implementation always selects the first UE created in script. Although MT can maximize cell throughput, it cannot provide fairness to UEs in poor channel condition.

Throughput to Average (TTA) Scheduler¶

The Throughput to Average (TTA) scheduler [FCapo2012] can be considered as an intermediate between MT and PF. The metric used in TTA is calculated as follows:

\widehat{i}_{k}(t) = \underset{j=1,...,N}{\operatorname{argmax}}

\left( \frac{ R_{j}(k,t) }{ R_{j}(t) } \right)

Here, R_{i}(k,t) in bit/s represents the achievable rate for user i on resource block k at subframe t. The calculation method already is shown in MT and PF. Meanwhile, R_{i}(t) in bit/s stands for the achievable rate for i at subframe t. The difference between those two achievable rates is how to get MCS. For R_{i}(k,t), MCS is calculated by subband CQI while R_{i}(t) is calculated by wideband CQI. TTA scheduler can only be implemented in frequency domain (FD) because the achievable rate of particular RBG is only related to FD scheduling.

Blind Average Throughput Scheduler¶

The Blind Average Throughput scheduler [FCapo2012] aims to provide equal throughput to all UEs under eNB. The metric used in TTA is calculated as follows:

\widehat{i}_{k}(t) = \underset{j=1,...,N}{\operatorname{argmax}}

\left( \frac{ 1 }{ T_\mathrm{j}(t) } \right)

where T_{j}(t) is the past throughput performance perceived by the user j and can be calculated by the same method in PF scheduler. In the time domain blind average throughput (TD-BET), the scheduler selects the UE with largest priority metric and allocates all RBGs to this UE. On the other hand, in the frequency domain blind average throughput (FD-BET), every TTI, the scheduler first selects one UE with lowest pastAverageThroughput (largest priority metric). Then scheduler assigns one RBG to this UE, it calculates expected throughput of this UE and uses it to compare with past average throughput T_{j}(t) of other UEs. The scheduler continues to allocate RBG to this UE until its expected throughput is not the smallest one among past average throughput T_{j}(t) of all UE. Then the scheduler will use the same way to allocate RBG for a new UE which has the lowest past average throughput T_{j}(t) until all RBGs are allocated to UEs. The principle behind this is that, in every TTI, the scheduler tries the best to achieve the equal throughput among all UEs.

Token Bank Fair Queue Scheduler¶

Token Bank Fair Queue (TBFQ) is a QoS aware scheduler which derives from the leaky-bucket mechanism. In TBFQ, a traffic flow of user i is characterized by following parameters:

Each K bytes data consumes k tokens. Also, TBFQ maintains a shared token bank (B) so as to balance the traffic between different flows. If token generation rate r_{i} is bigger than packet arrival rate t_{i}, then tokens overflowing from token pool are added to the token bank, and E_{i} is increased by the same amount. Otherwise, flow i needs to withdraw tokens from token bank based on a priority metric frac{E_{i}}{r_{i}}, and E_{i} is decreased. Obviously, the user contributes more on token bank has higher priority to borrow tokens; on the other hand, the user borrows more tokens from bank has lower priority to continue to withdraw tokens. Therefore, in case of several users having the same token generation rate, traffic rate and token pool size, user suffers from higher interference has more opportunity to borrow tokens from bank. In addition, TBFQ can police the traffic by setting the token generation rate to limit the throughput. Additionally, TBFQ also maintains following three parameters for each flow:

LTE in NS-3 has two versions of TBFQ scheduler: frequency domain TBFQ (FD-TBFQ) and time domain TBFQ (TD-TBFQ). In FD-TBFQ, the scheduler always select UE with highest metric and allocates RBG with highest subband CQI until there are no packets within UE’s RLC buffer or all RBGs are allocated [FABokhari2009]. In TD-TBFQ, after selecting UE with maximum metric, it allocates all RBGs to this UE by using wideband CQI [WKWong2004].

Priority Set Scheduler¶

Priority set scheduler (PSS) is a QoS aware scheduler which combines time domain (TD) and frequency domain (FD) packet scheduling operations into one scheduler [GMonghal2008]. It controls the fairness among UEs by a specified Target Bit Rate (TBR).

In TD scheduler part, PSS first selects UEs with non-empty RLC buffer and then divide them into two sets based on the TBR:

\widehat{i}_{k}(t) = \underset{j=1,...,N}{\operatorname{argmax}}

\left( \frac{ 1 }{ T_\mathrm{j}(t) } \right)

\widehat{i}_{k}(t) = \underset{j=1,...,N}{\operatorname{argmax}}

\left( \frac{ R_{j}(k,t) }{ T_\mathrm{j}(t) } \right)

UEs belonged to set 1 have higher priority than ones in set 2. Then PSS will select N_{mux} UEs with highest metric in two sets and forward those UE to FD scheduler. In PSS, FD scheduler allocates RBG k to UE n that maximums the chosen metric. Two PF schedulers are used in PF scheduler:

\widehat{Msch}_{k}(t) = \underset{j=1,...,N}{\operatorname{argmax}}

\left( \frac{ R_{j}(k,t) }{ Tsch_\mathrm{j}(t) } \right)

\widehat{Mcoi}_{k}(t) = \underset{j=1,...,N}{\operatorname{argmax}}

\left( \frac{ CoI[j,k] }{ \sum_{k=0}^{N_{RBG}} CoI[j,k] } \right)

where Tsch_{j}(t) is similar past throughput performance perceived by the user j, with the difference that it is updated only when the i-th user is actually served. CoI[j,k] is an estimation of the SINR on the RBG k of UE j. Both PFsch and CoIta is for decoupling FD metric from TD scheduler. In addition, PSS FD scheduler also provide a weight metric W[n] for helping controlling fairness in case of low number of UEs.

W[n] = max (1, \frac{TBR}{ T_{j}(t) })

where T_{j}(t) is the past throughput performance perceived by the user j . Therefore, on RBG k, the FD scheduler selects the UE j that maximizes the product of the frequency domain metric (Msch, MCoI) by weight W[n]. This strategy will guarantee the throughput of lower quality UE tend towards the TBR.

Config::SetDefault ("ns3::PfFfMacScheduler::HarqEnabled", BooleanValue (false));

The scheduler implements the filtering of the uplink CQIs according to their nature with UlCqiFilter attribute, in detail:

Channel and QoS Aware Scheduler¶

The Channel and QoS Aware (CQA) Scheduler [Bbojovic2014] is an LTE MAC downlink scheduling algorithm that considers the head of line (HOL) delay, the GBR parameters and channel quality over different subbands. The CQA scheduler is based on joint TD and FD scheduling.

In the TD (at each TTI) the CQA scheduler groups users by priority. The purpose of grouping is to enforce the FD scheduling to consider first the flows with highest HOL delay. The grouping metric m_{td} for user j=1,...,N is defined in the following way:

m_{td}^{j}(t) = \lceil\frac{d_{hol}^{j}(t)}{g}\rceil \;,

where d_{hol}^{j}(t) is the current value of HOL delay of flow j, and g is a grouping parameter that determines granularity of the groups, i.e. the number of the flows that will be considered in the FD scheduling iteration.

The groups of flows selected in the TD iteration are forwarded to the FD scheduling starting from the flows with the highest value of the m_{td} metric until all RBGs are assigned in the corresponding TTI. In the FD, for each RBG k=1,...,K, the CQA scheduler assigns the current RBG to the user j that has the maximum value of the FD metric which we define in the following way:

m_{fd}^{(k,j)}(t) = d_{HOL}^{j}(t) \cdot m_{GBR}^j(t) \cdot m_{ca}^{k,j}(t) \;,

where m_{GBR}^j(t) is calculated as follows:

m_{GBR}^j(t)=\frac{GBR^j}{\overline{R^j}(t)}=\frac{GBR^j}{(1-\alpha)\cdot\overline{R^j}(t-1)+\alpha \cdot r^j(t)} \;,

where GBR^j is the bit rate specified in EPS bearer of the flow j, \overline{R^j}(t) is the past averaged throughput that is calculated with a moving average, r^{j}(t) is the throughput achieved at the time t, and \alpha is a coefficient such that 0 \le \alpha \le1.

For m_{ca}^{(k,j)}(t) we consider two different metrics: m_{pf}^{(k,j)}(t) and m_{ff}^{(k,j)}(t). m_{pf} is the Proportional Fair metric which is defined as follows:

m_{pf}^{(k,j)}(t) = \frac{R_e^{(k,j)}}{\overline{R^j}(t)} \;,

where R_e^{(k,j)}(t) is the estimated achievable throughput of user j over RBG k calculated by the Adaptive Modulation and Coding (AMC) scheme that maps the channel quality indicator (CQI) value to the transport block size in bits.

The other channel awareness metric that we consider is m_{ff} which is proposed in [GMonghal2008] and it represents the frequency selective fading gains over RBG k for user j and is calculated in the following way:

m_{ff}^{(k,j)}(t) = \frac{CQI^{(k,j)}(t)}{\sum_{k=1}^{K}CQI(t)^{(k,j)}} \;,

where CQI^{(k,j)}(t) is the last reported CQI value from user j for the k-th RBG.

The user can select whether m_{pf} or m_{ff} is used by setting the attribute ns3::CqaFfMacScheduler::CqaMetric respectively to "CqaPf" or "CqaFf".

Random Access¶

The LTE model includes a model of the Random Access procedure based on some simplifying assumptions, which are detailed in the following for each of the messages and signals described in the specs [TS36321].

Figure Sequence diagram of the Contention-based MAC Random Access procedure and Sequence diagram of the Non-contention-based MAC Random Access procedure shows the sequence diagrams of respectively the contention-based and non-contention-based MAC random access procedure, highlighting the interactions between the MAC and the other entities.

RLC¶

Overview¶

The RLC entity is specified in the 3GPP technical specification [TS36322], and comprises three different types of RLC: Transparent Mode (TM), Unacknowledged Mode (UM) and Acknowledged Mode (AM). The simulator includes one model for each of these entities

The RLC entities provide the RLC service interface to the upper PDCP layer and the MAC service interface to the lower MAC layer. The RLC entities use the PDCP service interface from the upper PDCP layer and the MAC service interface from the lower MAC layer.

Figure Implementation Model of PDCP, RLC and MAC entities and SAPs shows the implementation model of the RLC entities and its relationship with all the other entities and services in the protocol stack.

Service Interfaces¶

RLC Service Interface¶

The RLC service interface is divided into two parts:

Both the UM and the AM RLC entities provide the same RLC service interface to the upper PDCP layer.

RLC Service Primitives¶

The following list specifies which service primitives are provided by the RLC service interfaces: