How to Begin Implement Voice over IP in NS3

To create a Voice over IP (VoIP) in NS-3 has been includes their replicate a together signaling and data transmission for voice communication over IP networks. This contains the launch a session for using the signaling protocol for instance SIP or a simplified version and transmitting voice packets using RTP (Real-Time Protocol).

Here’s a step-by-step guide:

Steps to Begin Implement Voice over IP in NS3

Step 1: Understand VoIP Requirements

1. Core Components:
   • Signaling Protocol: Use a protocol such as SIP to configure, handle, and terminate calls.
   • Media Transmission: Use to communicate the voice packets such as RTP or UDP.
   • QoS Metrics: Replicate the packet delay, jitter, and loss, which affect the voice quality.
2. Simplified VoIP Workflow:
   • Signaling phase: Introduce the session using SIP-like communications.
   • Media phase: Alter the voice packets among endpoints.
   • Termination phase: Finish the session.

Step 2: Set Up NS-3

1. Install NS-3:

git clone https://gitlab.com/nsnam/ns-3-dev.git

cd ns-3-dev

./ns3 configure –enable-examples –enable-tests

./ns3 build

1. Verify Installation: Execute a basic replication:

./ns3 run examples/tutorial/first

Step 3: Plan VoIP Implementation

1. Key Components:
   • VoIP Server: Maintain the signaling for session management.
   • VoIP Client: Recruits and receives the calls, and transmit the voice packets.
   • Voice Packet Simulation: It replicates the periodic generation for voice packets for instance 20 ms intervals.
2. Design Decisions:
   • Use UDP sockets for illustrate the transmitting RTP-like packets.
   • Concentrate on signaling and data variation rather than full SIP and RTP executions.

Step 4: ImplementVoIP Components

Step 4.1: Define the VoIP Server

Develop a class for the server application:

#include “ns3/application.h”

#include “ns3/socket.h”

#include “ns3/address.h”

using namespace ns3;

class VoipServer : public Application {

public:

static TypeId GetTypeId(void);

VoipServer();

virtual ~VoipServer();

private:

virtual void StartApplication() override;

virtual void StopApplication() override;

void HandleRequest(Ptr<Socket> socket);

Ptr<Socket> m_socket;

std::map<std::string, Address> m_registeredClients; // Registered clients

};

Step 4.2: Define the VoIP Client

Make a class for the client application:

#include “ns3/application.h”

#include “ns3/socket.h”

using namespace ns3;

class VoipClient : public Application {

public:

static TypeId GetTypeId(void);

VoipClient();

virtual ~VoipClient();

void SetServer(Address serverAddress);

private:

virtual void StartApplication() override;

virtual void StopApplication() override;

void RegisterWithServer();

void InitiateCall();

void SendVoicePackets();

void ReceiveVoicePackets(Ptr<Socket> socket);

Ptr<Socket> m_socket;

Address m_serverAddress;

Address m_peerAddress; // Call peer

};

Step 4.3: Implement Core Functions

• VoIP Server Logic:
  • Manage the SIP-like signaling and handle the client registrations.

void VoipServer::StartApplication() {

m_socket = Socket::CreateSocket(GetNode(), TypeId::LookupByName(“ns3::UdpSocketFactory”));

InetSocketAddress local = InetSocketAddress(Ipv4Address::GetAny(), 5060); // SIP Port

m_socket->Bind(local);

m_socket->SetRecvCallback(MakeCallback(&VoipServer::HandleRequest, this));

}

void VoipServer::StopApplication() {

if (m_socket) {

m_socket->Close();

}

}

void VoipServer::HandleRequest(Ptr<Socket> socket) {

Ptr<Packet> packet = socket->Recv();

std::string message = std::string((char *)packet->PeekData(), packet->GetSize());

std::cout << “Received: ” << message << std::endl;

// Handle REGISTER, INVITE, and BYE messages

if (message.find(“REGISTER”) == 0) {

std::string client = message.substr(9);

m_registeredClients[client] = socket->GetPeerName();

socket->Send(Create<Packet>((uint8_t *)”200 OK”, 6));

} else if (message.find(“INVITE”) == 0) {

// Handle INVITE and send acknowledgment

socket->Send(Create<Packet>((uint8_t *)”200 OK”, 6));

}

}

• VoIP Client Logic:
  • Record by the server, initiate a call, and send/receive voice packets.

void VoipClient::StartApplication() {

m_socket = Socket::CreateSocket(GetNode(), TypeId::LookupByName(“ns3::UdpSocketFactory”));

m_socket->Connect(m_serverAddress);

Simulator::Schedule(Seconds(1.0), &VoipClient::RegisterWithServer, this);

}

void VoipClient::StopApplication() {

if (m_socket) {

m_socket->Close();

}

}

void VoipClient::RegisterWithServer() {

Ptr<Packet> packet = Create<Packet>((uint8_t *)”REGISTER UA1″, 12);

m_socket->Send(packet);

}

void VoipClient::InitiateCall() {

Ptr<Packet> packet = Create<Packet>((uint8_t *)”INVITE UA2″, 10);

m_socket->Send(packet);

}

void VoipClient::SendVoicePackets() {

Ptr<Packet> packet = Create<Packet>(100); // Simulate voice data

m_socket->Send(packet);

Simulator::Schedule(MilliSeconds(20), &VoipClient::SendVoicePackets, this); // 20 ms intervals

}

void VoipClient::ReceiveVoicePackets(Ptr<Socket> socket) {

Ptr<Packet> packet = socket->Recv();

std::cout << “Received voice packet of size: ” << packet->GetSize() << ” bytes” << std::endl;

}

Step 5: Simulate VoIP Communication

Simulation Script:

Generate a basic topology by a VoIP server and two clients.

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

NodeContainer nodes;

nodes.Create(3); // 1 server, 2 clients

PointToPointHelper p2p;

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

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

NetDeviceContainer devices = p2p.Install(nodes.Get(0), nodes.Get(1));

devices.Add(p2p.Install(nodes.Get(0), nodes.Get(2)));

InternetStackHelper stack;

stack.Install(nodes);

Ipv4AddressHelper address;

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

Ipv4InterfaceContainer interfaces = address.Assign(devices);

// Set up VoIP Server

Ptr<VoipServer> server = CreateObject<VoipServer>();

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

server->SetStartTime(Seconds(1.0));

server->SetStopTime(Seconds(20.0));

// Set up VoIP Client 1

Ptr<VoipClient> client1 = CreateObject<VoipClient>();

client1->SetServer(InetSocketAddress(interfaces.GetAddress(0), 5060));

nodes.Get(1)->AddApplication(client1);

client1->SetStartTime(Seconds(2.0));

client1->SetStopTime(Seconds(20.0));

// Set up VoIP Client 2

Ptr<VoipClient> client2 = CreateObject<VoipClient>();

client2->SetServer(InetSocketAddress(interfaces.GetAddress(0), 5060));

nodes.Get(2)->AddApplication(client2);

client2->SetStartTime(Seconds(2.0));

client2->SetStopTime(Seconds(20.0));

Simulator::Run();

Simulator::Destroy();

return 0;

}

Step 6: Test and Debug

1. Enable Logging:

export NS_LOG=”VoipServer=level_all|prefix_time,VoipClient=level_all|prefix_time”

./ns3 run my-voip-simulation

1. Validate Results:
   • Validate the registering and launch the session.
   • Check for reliable the voice packet altercation.

Step 7: Extend and Optimize

1. Add Features:
   • Encourage the QoS parameter metrics such as delay, jitter, and packet loss.
   • Replicate the codec behavior for instance G.711, G.729.
2. Performance Testing:
   • It replicates the larger networks through numerous VoIP clients.
   • Use for congestion analysis FlowMonitor.

In this demonstration, we thoroughly understood on how to simulate and prolong the voice over IP projects across the ns3 analysis tool. If you did like to know more details concerning this process we will presented it.