How to Start Voice over IP Projects Using NS3

To start a Voice over IP (VoIP) project in NS3 that encompasses to replicate the real-time transmission of voice data across an IP network. Since NS3 doesn’t support built-in VoIP applications then their flexibility permits to replicate the VoIP traffic and also examine performance parameters like delay, jitter, packet loss, and MOS (Mean Opinion Score).

Following is a detailed guide to execute and simulate the VoIP Projects using NS3:

Steps to Start Voice over IP Projects in NS3

  1. Understand VoIP and Its Simulation Requirements
  • What is VoIP?
    • A technology, which allows voice interaction over IP networks to utilize protocols such as RTP, SIP, and RTCP.
    • For lightweight, real-time delivery, it functions across UDP.
  • Key Aspects to Simulate:
    • Voice encoding and packetization such as G.711, G.729.
    • Real-time transport to utilize RTP/UDP.
    • Performance metrics like delay, jitter, packet loss, MOS.
  1. Set Up NS3
  • Install NS3:
    • Go to nsnam.org to download and install NS3 environment.
    • We adhere to the installation instruction.
  • Verify Installation:

./waf –run scratch/test-example

  • Required Modules:
    • internet: It supports for IP and UDP functionalities.
    • applications: For custom or traffic-generating applications utilised these module.
    • point-to-point or wifi: For network topology.
  1. Plan Your VoIP Simulation
  • Topology:
    • A basic two-node configuration for VoIP interaction:
      • Node A (VoIP sender) → Node B (VoIP receiver).
    • Optional: Insert intermediate routers or several senders and receivers.
  • Simulation Goals:
    • To replicate the VoIP packet generation, transmission, and reception.
    • Examine network performance for real-time traffic.
  1. Create a Custom VoIP Application

VoIP normally contains RTP for real-time transport. While NS3 environment doesn’t support built-in RTP/VoIP executions, we will require replicating it with the help of custom applications.

4.1 Steps to Create a Custom VoIP Application

  1. Define VoIP Sender and Receiver Classes:
    • We can prolong the Application class, making VoipSenderApplication and VoipReceiverApplication.
  2. Simulate Voice Packetization:
    • Make packets at intervals like 20ms including the payload sizes based on the codec (e.g., 160 bytes for G.711).
  3. Handle UDP Communication:
    • For packet transmission and reception we can utilize UDP sockets.
  4. Monitor Performance:
    • Estimate the performance metrics like delay, jitter, and packet loss.

4.2 Example: Basic VoIP Sender and Receiver

VoIP Sender

#include “ns3/core-module.h”

#include “ns3/network-module.h”

#include “ns3/internet-module.h”

#include “ns3/applications-module.h”

using namespace ns3;

class VoipSender : public Application {

public:

VoipSender();

virtual ~VoipSender();

void Setup(Address address, uint16_t payloadSize, double interval);

private:

virtual void StartApplication();

virtual void StopApplication();

void SendPacket();

Ptr<Socket> m_socket;

Address     m_peerAddress;

uint16_t    m_payloadSize;

EventId     m_sendEvent;

double      m_interval; // Time interval between packets

};

VoipSender::VoipSender() : m_socket(0) {}

VoipSender::~VoipSender() {

m_socket = 0;

}

void VoipSender::Setup(Address address, uint16_t payloadSize, double interval) {

m_peerAddress = address;

m_payloadSize = payloadSize;

m_interval = interval;

}

void VoipSender::StartApplication() {

m_socket = Socket::CreateSocket(GetNode(), UdpSocketFactory::GetTypeId());

m_socket->Connect(m_peerAddress);

m_sendEvent = Simulator::Schedule(Seconds(0.0), &VoipSender::SendPacket, this);

}

void VoipSender::StopApplication() {

if (m_socket) {

m_socket->Close();

m_socket = 0;

}

Simulator::Cancel(m_sendEvent);

}

void VoipSender::SendPacket() {

Ptr<Packet> packet = Create<Packet>(m_payloadSize); // Simulated voice payload

m_socket->Send(packet);

Simulator::Schedule(Seconds(m_interval), &VoipSender::SendPacket, this);

}

VoIP Receiver

class VoipReceiver : public Application {

public:

VoipReceiver();

virtual ~VoipReceiver();

void Setup(Address address);

private:

virtual void StartApplication();

virtual void StopApplication();

void HandleRead(Ptr<Socket> socket);

Ptr<Socket> m_socket;

Address     m_local;

};

VoipReceiver::VoipReceiver() : m_socket(0) {}

VoipReceiver::~VoipReceiver() {

m_socket = 0;

}

void VoipReceiver::Setup(Address address) {

m_local = address;

}

void VoipReceiver::StartApplication() {

m_socket = Socket::CreateSocket(GetNode(), UdpSocketFactory::GetTypeId());

m_socket->Bind(m_local);

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

}

void VoipReceiver::StopApplication() {

if (m_socket) {

m_socket->Close();

m_socket = 0;

}

}

void VoipReceiver::HandleRead(Ptr<Socket> socket) {

Ptr<Packet> packet;

Address from;

while ((packet = socket->RecvFrom(from))) {

NS_LOG_INFO(“Received VoIP packet of size: ” << packet->GetSize());

}

}

  1. Simulate VoIP Communication

Main Simulation Script

int main(int argc, char *argv[]) {

NodeContainer nodes;

nodes.Create(2);

PointToPointHelper p2p;

p2p.SetDeviceAttribute(“DataRate”, StringValue(“10Mbps”));

p2p.SetChannelAttribute(“Delay”, StringValue(“2ms”));

NetDeviceContainer devices = p2p.Install(nodes);

InternetStackHelper stack;

stack.Install(nodes);

Ipv4AddressHelper ipv4;

ipv4.SetBase(“10.1.1.0”, “255.255.255.0”);

Ipv4InterfaceContainer interfaces = ipv4.Assign(devices);

// VoIP sender setup

Ptr<VoipSender> sender = CreateObject<VoipSender>();

sender->Setup(InetSocketAddress(interfaces.GetAddress(1), 12345), 160, 0.02); // 160 bytes, 20ms interval

nodes.Get(0)->AddApplication(sender);

sender->SetStartTime(Seconds(1.0));

sender->SetStopTime(Seconds(10.0));

// VoIP receiver setup

Ptr<VoipReceiver> receiver = CreateObject<VoipReceiver>();

receiver->Setup(InetSocketAddress(Ipv4Address::GetAny(), 12345));

nodes.Get(1)->AddApplication(receiver);

receiver->SetStartTime(Seconds(0.5));

receiver->SetStopTime(Seconds(10.0));

Simulator::Run();

Simulator::Destroy();

return 0;

}

  1. Test and Debug
  • Enable Logging:

export NS_LOG=VoipSender=level_all:VoipReceiver=level_all

./waf –run scratch/voip-simulation

  • Inspect Packet Flows:
    • Examine traffic in Wireshark to utilize PCAP tracing

p2p.EnablePcapAll(“voip”);

  1. Evaluate VoIP Performance
  • Metrics to Measure:
    • Latency: Measure the duration among packet transmission and reception.
    • Jitter: Variability within packet inter-arrival times.
    • Packet Loss: We compute the rate of packets that are dropped.
    • MOS (Mean Opinion Score): Determine perceived quality according to the delay, jitter, and loss.
  • Use FlowMonitor:

FlowMonitorHelper flowMonitor;

Ptr<FlowMonitor> monitor = flowMonitor.InstallAll();

monitor->SerializeToXmlFile(“voip-performance.xml”, true, true);

  1. Advanced Features
  • Codec Simulation:
    • We replicate the codecs such as G.711 (160 bytes at 20ms) or G.729 (20 bytes at 20ms).
  • Multicast VoIP:
    • We should insert multicast support for conferencing.
  • Simulate SIP for Call Setup:
    • Manage signaling to utilize a custom SIP application.
  • Introduce Network Conditions:
    • Append delays, packet loss, or bandwidth constraints replicating the real-world situations.

In this given module, we had explicitly focussed the advanced information and simple procedure on how to simulate and measure the Voice over IP Projects using NS3 tool. If you need more information regarding this process we will offer it based on your requirements.

If you’re interested in more updates on Voice over IP projects, feel free to contact us for assistance. By partnering with phdprojects.org, you can streamline your research and writing process, as we handle these challenging tasks for you. We focus on key performance metrics such as delay, jitter, packet loss, and MOS (Mean Opinion Score) tailored to your needs.