To simulate Voice over IP (VoIP) projects in OMNeT++ has includes to design the signalling protocols such as SIP and the media transport protocols like RTP utilized to introduce and handle voice communication over IP networks. VoIP simulations enabling you to measure the performance of voice communication based on latency, jitter, packet loss, and overall call quality.
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Here’s a step-by-step guide to simulate a VoIP system in OMNeT++ using the INET framework:
Key Concepts for VoIP Simulation
- SIP (Session Initiation Protocol): Used for signalling to introduce, maintain, and stop voice sessions.
- RTP (Real-Time Transport Protocol): Utilized for transporting the voice data (media) in real-time.
- QoS (Quality of Service): parameters like delay, jitter, and packet loss are significant for measuring call quality.
Steps to Simulate Voice over IP Projects Using OMNeT++
Step 1: Install OMNeT++ and INET Framework
Make sure that OMNeT++ and the INET framework are installed on the system. INET provides support for necessary network components such as hosts, routers, and communication channels that we can utilize to replicate VoIP interaction.
- Download OMNeT++: OMNeT++
- Download INET Framework: INET Framework
Step 2: Define the Network Topology in the NED File
Describe a network topology in a .ned file that contains VoIP clients (endpoints) and VoIP servers (such as SIP proxy and RTP media server). The clients will begin and receive VoIP calls, since the servers manage call signalling and media routing.
Here’s a basic example of a VoIP network:
network VoIPNetwork
{
submodules:
voipClient1: StandardHost {
@display(“p=100,200”);
}
voipClient2: StandardHost {
@display(“p=500,200”);
}
sipProxy: StandardHost {
@display(“p=300,300”);
}
connections:
voipClient1.ethg++ <–> Eth100M <–> sipProxy.ethg++;
voipClient2.ethg++ <–> Eth100M <–> sipProxy.ethg++;
}
In this topology:
- voipClient1 and voipClient2 signify two VoIP endpoints that will interact with each other.
- sipProxy is a SIP proxy server that manage signalling among the two clients.
Step 3: Implement SIP for Call Signaling
SIP is utilized to start, adapt, and stop VoIP sessions. We can replicate the SIP messages interchanged among the VoIP clients and the SIP proxy server.
Step 3.1: SIP Client
The SIP client will transmit INVITE messages to begin a VoIP session and process responses such as TRYING, RINGING, and OK. Once the session is introduced, the client can transmit RTP packets for the voice communication.
class VoIPClient : public cSimpleModule
{
protected:
virtual void initialize() override;
virtual void handleMessage(cMessage *msg) override;
void sendInvite();
void processSIPResponse(cMessage *msg);
void startRTPStream();
};
Define_Module(VoIPClient);
void VoIPClient::initialize()
{
// Send INVITE message to start a call at the beginning of the simulation
if (strcmp(getName(), “voipClient1”) == 0) {
sendInvite();
}
}
void VoIPClient::handleMessage(cMessage *msg)
{
processSIPResponse(msg);
}
void VoIPClient::sendInvite()
{
// Send an INVITE request to the SIP proxy
EV << “Sending INVITE from voipClient1 to sipProxy\n”;
cMessage *invite = new cMessage(“INVITE”);
send(invite, “ethg$o”);
}
void VoIPClient::processSIPResponse(cMessage *msg)
{
if (strcmp(msg->getName(), “100 TRYING”) == 0) {
EV << “Received TRYING response\n”;
} else if (strcmp(msg->getName(), “180 RINGING”) == 0) {
EV << “Received RINGING response\n”;
} else if (strcmp(msg->getName(), “200 OK”) == 0) {
EV << “Received OK, starting RTP stream\n”;
startRTPStream();
}
delete msg;
}
void VoIPClient::startRTPStream()
{
// Start sending RTP packets for the voice stream
cMessage *rtpPacket = new cMessage(“RTP Voice Packet”);
send(rtpPacket, “ethg$o”);
}
- sendInvite() sends an INVITE message to begins the call.
- processSIPResponse() processes responses such as TRYING, RINGING, and OK. When OK is received, it begins to sending RTP packets for the voice stream.
Step 3.2: SIP Proxy Server
The SIP proxy server processes the INVITE message from the caller and forwards it to the callee. It transmits TRYING and RINGING responses back to the caller and manage call termination with BYE.
class SIPProxy : public cSimpleModule
{
protected:
virtual void handleMessage(cMessage *msg) override;
void processInvite(cMessage *msg);
};
Define_Module(SIPProxy);
void SIPProxy::handleMessage(cMessage *msg)
{
if (strcmp(msg->getName(), “INVITE”) == 0) {
processInvite(msg);
}
delete msg;
}
void SIPProxy::processInvite(cMessage *msg)
{
EV << “Proxy received INVITE, forwarding to voipClient2\n”;
// Send TRYING response back to voipClient1
cMessage *trying = new cMessage(“100 TRYING”);
send(trying, “ethg$o”, 0); // Send to voipClient1
// Simulate forwarding INVITE to voipClient2 and getting a response
cMessage *ringing = new cMessage(“180 RINGING”);
send(ringing, “ethg$o”, 1); // Send to voipClient2
}
The proxy server manages the signalling interchange and replicate responses back to the clients.
Step 4: Implement RTP for Media Transmission
Once the SIP session is introduced, RTP (Real-Time Transport Protocol) is utilized to carry the voice data. We can replicate RTP packet transmission among the VoIP clients.
void VoIPClient::startRTPStream()
{
// Start sending RTP packets at regular intervals
cMessage *rtpPacket = new cMessage(“RTP Voice Packet”);
scheduleAt(simTime() + 0.02, rtpPacket); // Send every 20 ms
}
void VoIPClient::handleMessage(cMessage *msg)
{
if (strcmp(msg->getName(), “RTP Voice Packet”) == 0) {
EV << “Sending RTP packet\n”;
send(msg, “ethg$o”);
// Schedule the next RTP packet
scheduleAt(simTime() + 0.02, msg); // 20 ms interval
} else {
processSIPResponse(msg);
}
}
In this example:
- RTP packets are transmit every 20 milliseconds to replicate real-time voice transmission.
Step 5: Configure the Simulation in omnetpp.ini
In the omnetpp.ini file, set up the network settings that involve IP addresses for the clients and the SIP proxy server, as well as link parameters like data rate and delay.
network = VoIPNetwork
sim-time-limit = 100s
# Assign IP addresses to the VoIP clients and SIP proxy
*.voipClient1.ipv4.config = “manual”
*.voipClient1.ipv4.addresses = “10.0.0.1/24”
*.voipClient2.ipv4.config = “manual”
*.voipClient2.ipv4.addresses = “10.0.0.2/24”
*.sipProxy.ipv4.config = “manual”
*.sipProxy.ipv4.addresses = “10.0.0.254/24”
# Set up link parameters (e.g., delay, bandwidth)
*.voipClient1.eth[0].datarate = 100Mbps
*.voipClient2.eth[0].datarate = 100Mbps
*.sipProxy.eth[0].datarate = 100Mbps
Step 6: Run the Simulation
After configure the network and execute the SIP and RTP protocols:
- Build and compile the project in OMNeT++.
- Run the simulation in the OMNeT++ GUI or from the command line.
- Observe the VoIP call: The SIP messages (INVITE, TRYING, RINGING, OK) will be interchange among the clients and the proxy server, followed by the RTP media packets.
Step 7: Analyse VoIP Metrics
OMNeT++ creates .sca and .vec files containing data that can utilize to measure the VoIP call’s performance:
- End-to-end delay: Evaluate the latency among when an RTP packet is sent and when it is received.
- Jitter: Assess the variation in the latency of RTP packets.
- Packet loss: measure the percentage of RTP packets that are lost because of network congestion or other factors.
- Call setup time: The time it takes to introduce the SIP session from INVITE to OK.
- Call quality: Utilizing parameters such as MOS (Mean Opinion Score) to measure perceived call quality.
Step 8: Extend the VoIP Simulation
We can expand the simulation by adding characteristics such as:
8.1: Introducing Network Delays and Packet Loss
Replicate network conditions that impact VoIP quality, like latency, packet loss, or limited bandwidth.
# Add delay and packet loss to the links
*.voipClient1.eth[0].delay = 10ms
*.sipProxy.eth[0].delay = 20ms
*.sipProxy.eth[0].lossRate = 0.05 # 5% packet loss
8.2: Implement QoS Mechanisms
Replicate Quality of Service (QoS) mechanisms, like selects VoIP traffic over other kinds of traffic or reserving bandwidth for voice communication.
8.3: Simulating Call Failures
Establish network failures or dropped packets that cause to SIP call termination or retry logic.
8.4: Adding Multiple VoIP Calls
Implement a scenario with multiple simultaneous VoIP calls to monitor how the network manages increased traffic and the effect on call quality.
Step 9: Advanced VoIP Features
- RTP over UDP/TCP: Incorporate transport layer protocols for RTP, replicates how RTP act as over UDP or TCP.
- SIP over TLS: Replicate secure SIP interaction using TLS (Transport Layer Security).
- VoIP with Media Servers: prolong the network to contain a media server that manages RTP packet forwarding or transcoding.
In this demonstration, we thoroughly understood on how to simulate and prolong the voice over IP projects across the OMNeT++ analysis tool. If you did like to know more details regarding this process we will offered it.
