How to Start Voice over IP Projects Using OMNeT++

To start a Voice over IP (VoIP) project using OMNeT++, we can replicate the voice data transmission over IP networks. This project normally has some protocols such as RTCP (Real-time Control Protocol) for observing the call’s quality, RTP (Real-time Transport Protocol) for voice data transport, and SIP (Session Initiation Protocol) for session configuration. We will walk you through step-by-step guide to get started:

Steps to Start VoIP Projects in OMNeT++

  1. Set Up OMNeT++ Environment

Download and Install OMNeT++

  1. We should download and install the OMNeT++ on the system.
  2. Adhere to the installation guidance based on OS.
  3. Confirm the installation with example simulations properly installed.

Install INET Framework

The INET is supports to replicate network protocols like IP, UDP, and TCP that are required for VoIP simulations.

  1. Make use of GitHub to download the INET framework.
  2. Construct and set the INET in OMNeT++.
  1. Understand VoIP Basics

VoIP normally has several modules like:

  • SIP (Session Initiation Protocol): It helps to manage the call configuration, modification, and termination.
  • RTP (Real-time Transport Protocol): It exports the voice information.
  • RTCP (Real-time Transport Control Protocol): It offers quality observing and feedback on the call like jitter, latency.

In VoIP, crucial challenges contain to make sure that low latency, low jitter, and high quality in the course of the voice conversation.

  1. Create a New OMNeT++ Project
  1. Create a Project:
    • Go to OMNeT++ and then choose File > New > OMNeT++ Project.
    • Name it to the project like VoIPProject.
  2. Link to INET Framework:
    • Right-click on the project, select Properties > Project References, and connect to the project to INET Framework.
  1. Design the Network Topology

We will want minimum two clients (User Agents or UAs) for a basic VoIP simulation and a VoIP server or a router for routing the packets. The simple network topology has to contain:

  • VoIP Clients (SIP-based): These devices transmit and obtain the voice information.
  • VoIP Server: This server can be managed SIP demands or only function like a routing intermediary.
  • Router: It handles the transmission of data among the clients as required.
  • Network Links: Offers the interaction infrastructure among the devices.

Below is a basic sample of a network topology within NED format:

Example VoIPNetwork.ned:

network VoIPNetwork {

submodules:

voipClient1: StandardHost {

@display(“p=100,100”);

}

voipClient2: StandardHost {

@display(“p=300,100”);

}

voipServer: StandardHost {

@display(“p=200,100”);

}

router: Router {

@display(“p=200,200”);

}

connections allowunconnected:

voipClient1.pppg++ <–> router.pppg++;

voipClient2.pppg++ <–> router.pppg++;

router.pppg++ <–> voipServer.pppg++;

}

In this example:

  • voipClient1 and voipClient2 are the VoIP clients, which establish and obtain calls.
  • voipServer performs like a SIP proxy or it can be utilised for routing.
  • router links the clients and the server.
  1. Implement SIP for Session Setup

Since OMNeT++ doesn’t directly support built-in SIP then we can make a simple SIP client and SIP server handling the call configuration and teardown. We can use SIP examples or adjust existing ones to make own execution in OMNeT++.

For instance, a SIP client could start a call by transmitting an INVITE message to the SIP server, and the SIP server responds including an OK or RINGING message. In the same way, the SIP client transmits a BYE message, ending the call.

Below is a basic skeleton of SIP client and server:

  • SIP Client: It transmits INVITE to start a call and BYE for terminating the call.
  • SIP Server: Pay attention for INVITE, replies with OK that manages call termination with BYE.
  1. Implement RTP for Voice Data Transport

RTP (Real-time Transport Protocol) is frequently utilised for sending voice information among the clients. We can be utilised UDP which is the transport layer for RTP packets.

We can prolong the UDP module for RTP implementation within OMNeT++ and then transmit audio packets among the SIP clients.

Example RTP Client Implementation:

  1. Make a module, which makes RTP packets that can be according to the prerecorded voice samples or synthesized data.
  2. Transmit the RTP packets to the other client through the server or directly to utilise UDP sockets.

Below is a sample skeleton of how to transmit RTP packets:

class RTPClient : public cSimpleModule {

protected:

cMessage *sendRTPMessage;

cUdpSocket *socket;

virtual void initialize() {

socket = new cUdpSocket();

sendRTPMessage = new cMessage(“SendRTP”);

scheduleAt(simTime() + 1.0, sendRTPMessage); // Start RTP transmission

}

virtual void handleMessage(cMessage *msg) {

if (msg == sendRTPMessage) {

// Send RTP packet (simplified example)

const char* rtpPacket = “RTP packet data”;  // In reality, this will be voice data

socket->sendTo(rtpPacket, strlen(rtpPacket), “voipClient2”, 5060);

}

}

virtual void finish() {

delete socket;

delete sendRTPMessage;

}

};

  1. Implement RTCP for Call Quality Monitoring

Real-time Transport Control Protocol (RTCP) observes the call’s quality by offering feedback regarding the voice quality like jitter, packet loss, and round-trip time. We can be executed a simple RTCP sender report and receiver report, replicating this aspect.

RTCP feedback messages are normally transmitted for each 5 seconds and have statistics on the performance of call.

Here is an instance RTCP message to manage:

class RTCPClient : public cSimpleModule {

protected:

cMessage *sendRTCPMessage;

virtual void initialize() {

sendRTCPMessage = new cMessage(“SendRTCP”);

scheduleAt(simTime() + 5.0, sendRTCPMessage); // Send RTCP feedback every 5 seconds

}

virtual void handleMessage(cMessage *msg) {

if (msg == sendRTCPMessage) {

// Send RTCP feedback (simplified)

const char* rtcpReport = “RTCP packet with call quality data”;

// In a real implementation, send actual data related to jitter, packet loss, etc.

send(rtcpReport, “voipClient2”);  // Send RTCP to the other client

}

}

virtual void finish() {

delete sendRTCPMessage;

}

};

  1. Configure Simulation Parameters in omnetpp.ini

Set the network simulation metrics, SIP settings, and allow debugging for SIP/RTP/RTCP messages using omnetpp.ini.

Example omnetpp.ini:

network = VoIPNetwork

sim-time-limit = 100s

# SIP Client Settings

*.voipClient1.sendInviteMessage = true

*.voipClient1.sendByeMessage = false

*.voipClient1.serverAddress = “voipServer”

*.voipClient1.port = 5060

# RTP Client Settings

*.voipClient1.rtpEnabled = true

*.voipClient1.rtpPort = 5004

# RTCP Client Settings

*.voipClient1.rtcpEnabled = true

# Enable debugging for SIP/RTP/RTCP communication

*.voipClient1.debug = true

*.voipServer.debug = true

  1. Run the Simulation
  • Make use of OMNeT++’s Qtenv or Tkenv graphical interface to establish the simulation.
  • In the simulation, we observe the SIP messages like INVITE, BYE, RTP packet transmission, and RTCP feedback.
  • Verify the interaction flow and performance to utilise debugging.
  1. Analyze and Optimize
  • Measure Call Setup Time: Estimate the duration for starting a VoIP call from initiation to acceptance.
  • Measure Packet Loss and Jitter: Make use of RTCP logs to estimate the quality of call such as jitter, latency, and packet loss.
  • Optimize for Performance: Experiment with diverse network conditions like diverse bandwidth, packet loss, replicating real-world performance of VoIP.
  1. Extend the Project

Prolong the project by means of integrating further aspects like:

  • SIP Authentication: Execute the SIP Digest Authentication to protect calls.
  • Multiple Clients: Replicate the calls among many clients within a network.
  • Quality of Service (QoS): Execute the QoS mechanisms to secure bandwidth for VoIP traffic.
  • NAT Traversal: We can execute the mechanisms for VoIP clients after NAT.

Step-by-step instructions and sample coding for replicating Voice over IP Projects were offered using OMNeT++ environment, with further insights to be incorporated into the next manual if required.

Let us manage your project’s performance! We expertly handle RTP (Real-time Transport Protocol) for voice data transmission, along with SIP (Session Initiation Protocol). Reach out to phdprojects.org for tailored research support, and we’ll be thrilled to assist you promptly.