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How to Implement Voice over IP in OMNeT++

To implement the Voice over IP (VoIP) in OMNeT++ has needs to emulate the whole process to setup, manage and terminating voice communication over an IP network. The VoIP systems usually use protocols such as Session Initiation Protocol (SIP) for call setup and Real-time Transport Protocol (RTP) for media transmission.  The given below are the step-by-procedure on how to implement the basic VoIP simulation in OMNeT++ using the INET framework.

Step-by-Step Implementation:

Step 1: Set Up the OMNeT++ Environment

  1. Install OMNeT++:
    • Make sure OMNeT++ is installed on system. Download it from the OMNeT++
  2. Install the INET Framework:
    • INET provide needed network protocol models such as TCP/UDP/IP, SIP, and RTP, which are essential for VoIP. We need to download it from the INET GitHub repository.

Step 2: Create a New OMNeT++ Project

  1. Create the Project:
    • Open OMNeT++ and create a new OMNeT++ project by selecting File > New > OMNeT++ Project.
    • Name project, for example, VoIP_Simulation, and set up the project directory.
  2. Set Up Project Dependencies:
    • Make sure that the project references the INET Framework by right-clicking on project in the Project Explorer, navigating to Properties > Project References, and checking the INET project.

Step 3: Understand the Components of VoIP

Before implementing VoIP, make clear yourself with the main components:

  • SIP (Session Initiation Protocol): To manage signalling for setting up, managing, and terminating VoIP calls.
  • RTP (Real-Time Transport Protocol): Carries the actual voice data.
  • RTCP (Real-Time Control Protocol): Provides feedback on the quality of the RTP stream.
  • SIP User Agents (UA): Devices or applications that can make and receive VoIP calls.
  • SIP Proxy Server: Routes SIP messages between UAs.
  • RTP Media Streams: Streams of audio data sent over the network using RTP.

Step 4: Define the Network Topology in OMNeT++

  1. Create a NED File:
    • Describe the network topology using the NED language. The topology should contain SIP clients (UAs), a SIP proxy server, and the network infrastructure like routers.

Example:

network VoIPNetwork

{

submodules:

client1: StandardHost;

client2: StandardHost;

sipProxy: StandardHost;

router: Router;

connections:

client1.ethg++ <–> Eth10Mbps <–> router.ethg++;

client2.ethg++ <–> Eth10Mbps <–> router.ethg++;

sipProxy.ethg++ <–> Eth10Mbps <–> router.ethg++;

}

  1. Configure Network Parameters:
    • Set up link parameters like bandwidth, delay, and packet loss to mimic the realistic network conditions for VoIP traffic.

Step 5: Implement SIP for Call Setup

  1. Define SIP Modules:
    • Create NED modules for the SIP clients and SIP proxy server.

Example (in NED):

simple SIPUserAgent

{

parameters:

@display(“i=block/cogwheel”);

gates:

inout udpIn;

inout udpOut;

}

simple SIPProxyServer

{

parameters:

@display(“i=block/server”);

gates:

inout udpIn;

inout udpOut;

}

  1. Implement SIP Logic in C++:
    • SIP User Agent Implementation:
      • Write the C++ code to manage SIP messages such as INVITE, ACK, BYE, and handle the call state.

Example (in C++):

void SIPUserAgent::handleMessage(cMessage *msg) {

if (msg->isSelfMessage()) {

initiateCall();

} else if (msg->arrivedOn(“udpIn”)) {

handleSIPMessage(msg);

}

}

void SIPUserAgent::initiateCall() {

SIPMessage *invite = new SIPMessage(“INVITE”);

invite->setCommand(“INVITE”);

invite->setUri(“sip:client2@domain.com”);

send(invite, “udpOut”);

}

void SIPUserAgent::handleSIPMessage(cMessage *msg) {

SIPMessage *sipMsg = check_and_cast<SIPMessage *>(msg);

if (sipMsg->getCommand() == “200 OK”) {

// Call setup complete, start RTP session

startRTPStream();

} else if (sipMsg->getCommand() == “BYE”) {

// Call termination

sendResponse(“200 OK”);

}

}

    • SIP Proxy Server Implementation:
      • To execute the SIP proxy server to route SIP messages among clients.

Example (in C++):

void SIPProxyServer::handleMessage(cMessage *msg) {

if (msg->arrivedOn(“udpIn”)) {

routeSIPMessage(msg);

}

}

void SIPProxyServer::routeSIPMessage(cMessage *msg) {

SIPMessage *sipMsg = check_and_cast<SIPMessage *>(msg);

if (sipMsg->getCommand() == “INVITE”) {

// Forward INVITE to the intended recipient

send(sipMsg, “udpOut”);

} else if (sipMsg->getCommand() == “REGISTER”) {

// Handle registration

send(sipMsg, “udpOut”);

}

}

Step 6: Implement RTP for Media Transmission

  1. Define RTP Modules:
    • Generate NED modules for the RTP sender and receiver.

Example (in NED):

simple RTPSender

{

parameters:

@display(“i=block/cogwheel”);

gates:

inout udpIn;

inout udpOut;

}

simple RTPReceiver

{

parameters:

@display(“i=block/cogwheel”);

gates:

inout udpIn;

inout udpOut;

}

  1. Implement RTP Logic in C++:
    • RTP Sender Implementation:
      • Write the C++ code to make RTP packets with voice data and send them over UDP.

Example (in C++):

void RTPSender::handleMessage(cMessage *msg) {

if (msg->isSelfMessage()) {

sendRTPPacket();

}

}

void RTPSender::sendRTPPacket() {

RTPPacket *packet = new RTPPacket(“RTPPacket”);

packet->setSequenceNumber(sequenceNumber++);

packet->setTimestamp(simTime().inUnit(SIMTIME_MS));

packet->setPayloadType(0);  // Assume payload type 0 for PCM audio

packet->setPayload(“VoiceData”);

send(packet, “udpOut”);

}

    • RTP Receiver Implementation:
      • In the simulation, write the C++ code to manage incoming RTP packets, process the voice data, and play it back.

Example (in C++):

void RTPReceiver::handleMessage(cMessage *msg) {

if (msg->arrivedOn(“udpIn”)) {

handleRTPPacket(check_and_cast<RTPPacket *>(msg));

}

}

void RTPReceiver::handleRTPPacket(RTPPacket *packet) {

int sequenceNumber = packet->getSequenceNumber();

simtime_t timestamp = packet->getTimestamp();

const char* payload = packet->getPayload();

EV << “Received RTP packet with sequence number ” << sequenceNumber << ” and timestamp ” << timestamp << endl;

 

delete packet;

}

Step 7: Configure the Simulation

  1. Set Up the UDP Stack:
    • In the omnetpp.ini file, configure the SIP and RTP components to use the UDP stack offered by INET.

Example:

network = VoIPNetwork

*.client1.numUdpApps = 2

*.client1.udpApp[0].typename = “UDPBasicApp”

*.client1.udpApp[0].destPort = 5060  // SIP uses port 5060

*.client1.udpApp[0].destAddress = “sipProxy”

*.client1.udpApp[1].typename = “UDPBasicApp”

*.client1.udpApp[1].destPort = 5004  // RTP uses port 5004

*.client1.udpApp[1].destAddress = “client2”

*.client2.numUdpApps = 2

*.client2.udpApp[0].typename = “UDPBasicApp”

*.client2.udpApp[0].localPort = 5060

*.client2.udpApp[1].typename = “UDPBasicApp”

*.client2.udpApp[1].localPort = 5004

*.sipProxy.numUdpApps = 1

*.sipProxy.udpApp[0].typename = “UDPBasicApp”

*.sipProxy.udpApp[0].localPort = 5060

  1. Define VoIP Scenarios:
    • Set up scenarios to test call setup, media transmission, and call termination under numerous network conditions.

Step 8: Simulate and Test VoIP

  1. Compile the Project:
    • Build OMNeT++ project to make sure everything is correctly executed.
  2. Run the Simulation:
    • To implement the simulation and monitor the SIP call setup, RTP media transmission, and call termination. Make sure that the complete VoIP communication process works as predictable.
  3. Analyse Results:
    • Use OMNeT++ tools to observe SIP and RTP message flows, measure quality metrics such as packet loss and jitter, and measure the overall performance of the VoIP system.

Step 9: Refine and Extend

  1. Enhance VoIP Functionality:
    • Add features such as support for multiple simultaneous calls, dynamic jitter buffers, and adaptive codecs.
  2. Test with Different Network Configurations:
    • To mimic the various network conditions like high latency, packet loss to validate the robustness of VoIP implementation.
  3. Implement RTCP for Feedback:
    • To execute RTCP to offer feedback on the quality of the RTP stream, like packet loss, delay, and jitter.

This module demonstrates how to setup and manage the voice communication over an IP network using the OMNeT++ simulator. Further details will be provided about how the VoIP will be performing in other simulation tool. We can help you with the best simulation results if you send us the information of your project. We implement voice communication over an IP network using the OMNeT++ simulator tool. You may get the greatest project performance support from omnet-manual.com.