How to Develop Session Initiation Protocol Projects Using NS3

To develop the Session Initiation Protocol (SIP) Projects using NS-3 involves simulate an Session Initiation Protocol (SIP) is signaling protocol used to launch, change, and decrease the multimedia sessions like as Voice over IP (VoIP), video calls, and instant messaging. In NS3, we can replicate the SIP-based communication among their clients and servers over an IP network.

Steps to Develop a SIP Project in NS3

  1. Setup NS3 Environment

We enable the tool NS3 is installed:

sudo apt update

sudo apt install git

git clone

cd ns-3-dev

./ns3 configure

./ns3 build

  1. Define SIP Network Topology

Generate a network that contains of:

  • RTP Media Server (for voice/video transport)
  • SIP Clients (User Agents – UA)
  • SIP Proxy Server
  • SIP Registrar
  • Routers and Links (for network communication)

Example Topology

[SIP Client 1] —- [Router] —- [SIP Proxy Server] —- [SIP Client 2]

  • SIP Clients: Send and receive the SIP communication (REGISTER, INVITE, ACK, BYE)
  • SIP Proxy Server: Routes for communicate the SIP among their clients
  • SIP Registrar: Secure the user information and validate the details
  • RTP (Real-time Transport Protocol) Server: Manage the actual media streaming
  1. Implement SIP Protocol in NS3

Since NS3 cannot have built-in SIP design, we can replicate the SIP signaling over TCP/UDP using applications and sockets.

(a) Simulate SIP Client-Server Communication

  1. TcpSocketFactory for used the SIP over TCP (Port 5060).
  2. Used for SIP over UDP (Port 5060) in UdpSocketFactory.
  3. Apply the execution for Packet Exchange among their SIP clients and servers.

(b) SIP Message Flow

The SIP session involves:

  1. REGISTER → The client registers by SIP server.
  2. INVITE → The client transmitting an invite request to additional the SIP user.
  3. TRYING → The server is handle the invite.
  4. RINGING → It recognize the rings.
  5. OK → The call is established.
  6. ACK → The meeting is confirmed.
  7. BYE → The gathering is terminated.
  1. NS-3 Code Implementation for SIP

(a) State the Network Topology of SIP

#include “ns3/core-module.h”

#include “ns3/network-module.h”

#include “ns3/internet-module.h”

#include “ns3/point-to-point-module.h”

#include “ns3/applications-module.h”

using namespace ns3;

int main () {

NodeContainer nodes;

nodes.Create (3); // SIP Client 1, Proxy Server, SIP Client 2

PointToPointHelper p2p;

p2p.SetDeviceAttribute (“DataRate”, StringValue (“100Mbps”));

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

NetDeviceContainer devices;

devices = p2p.Install(nodes.Get(0), nodes.Get(1)); // Client 1 to Server

devices = p2p.Install(nodes.Get(1), nodes.Get(2)); // Server to Client 2

InternetStackHelper internet;

internet.Install(nodes);

Ipv4AddressHelper ipv4;

ipv4.SetBase (“192.168.1.0”, “255.255.255.0”);

Ipv4InterfaceContainer interfaces = ipv4.Assign(devices);

// Define SIP Server Application

uint16_t port = 5060; // SIP Server Port

Address serverAddress(InetSocketAddress(interfaces.GetAddress(1), port));

PacketSinkHelper sipServer (“ns3::UdpSocketFactory”, serverAddress);

ApplicationContainer serverApp = sipServer.Install(nodes.Get(1));

serverApp.Start(Seconds(1.0));

serverApp.Stop(Seconds(10.0));

// Define SIP Client 1 Application

UdpClientHelper sipClient1(serverAddress, port);

sipClient1.SetAttribute (“MaxPackets”, UintegerValue (10));

sipClient1.SetAttribute (“Interval”, TimeValue (Seconds (1.0)));

sipClient1.SetAttribute (“PacketSize”, UintegerValue (512));

ApplicationContainer clientApp1 = sipClient1.Install(nodes.Get(0));

clientApp1.Start(Seconds(2.0));

clientApp1.Stop(Seconds(9.0));

Simulator::Run();

Simulator::Destroy();

return 0;

}

  1. Simulate RTP for Voice Transmission

SIP only manage the call setting, although RTP (Real-time Transport Protocol) maintain actual media transmission.

(a) Configure RTP Flow

  1. OnOffApplication used for RTP congestion.
  2. State the audio/video packet transmission over UDP.

(b) NS-3 Code for RTP Traffic

uint16_t rtpPort = 5004; // RTP Port

Address rtpServerAddress (InetSocketAddress(interfaces.GetAddress(2), rtpPort));

OnOffHelper rtpStream (“ns3::UdpSocketFactory”, rtpServerAddress);

rtpStream.SetConstantRate(DataRate(“64kbps”)); // Simulate voice bitrate

ApplicationContainer rtpApp = rtpStream.Install(nodes.Get(0));

rtpApp.Start(Seconds(3.0)); // RTP starts after SIP setup

rtpApp.Stop(Seconds(9.0));

  1. Performance Analysis

Use to calculate the performance of investigation is FlowMonitor:

  • SIP Call Setup Delay
  • RTP Packet Loss
  • Jitter and Latency
  • End-to-End Delay

FlowMonitorHelper flowmon;

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

monitor->SerializeToXmlFile(“sip-results.xml”, true, true);

  1. Visualization & Analysis
  • Use to envision the SIP call flow NetAnim.
  • Distribute the data to MATLAB/Python for further analysis.
  1. Advanced Extensions
  • SIP-based on VoIP Performance Comparison over UDP vs. TCP.
  • Effects of Network traffic on SIP call quality.
  • SIP replicate over the LTE/5G in NS-3.
  • SIP Security Analysis (SIP over TLS/DTLS).
  • SIP-based on Video Conferencing Replication.

In the conclusion, we can gain the valuable insights about how the Session Initiation Protocol example projects will implement in diverse scenarios by using the ns3 tool. We also deliver the additional information regarding the Session Initiation Protocol.