How to Start Session Initiation Protocol Projects Using NS3

To start a Session Initiation Protocol (SIP) project using NS3, we’ll need to replicate the SIP’s behaviour that is a signalling protocol utilized to start, alter, and finish the real-time interaction sessions such as voice, video calls, and instant messaging. Even though NS3 doesn’t have built-in SIP functionality then we can be replicated it by making custom applications or combining the external SIP frameworks.

Below is a simplified approach on how we can execute such protocol using NS3.

Steps to Start Session Initiation Protocol Projects in NS3

Understand SIP and Its Simulation Requirements

What is SIP?
- SIP is a signaling protocol described within RFC 3261 for handling the interaction sessions.
- It functions over UDP, TCP, or TLS for secure interactions.
- Key Messages:
  - INVITE: It starts a session.
  - ACK: Acknowledges session setup.
  - BYE: Ends a session.
  - OPTIONS: Queries capabilities.
  - REGISTER: Registers a client including a SIP server.
Key Aspects to Simulate:
- SIP signaling among the user agents (clients) and SIP servers.
- To start interaction sessions among the nodes.
- Performance parameters such as signaling delay, throughput, and packet loss.

Set Up NS3

Install NS3:

- Go to nsnam.org to download and install NS3.
- We adhere to the installation instruction.
Verify Installation:

./waf –run scratch/test-example

Required Modules:
- internet: It supports for IP and TCP/UDP interaction.
- applications: This module used for making SIP-like custom applications.
- point-to-point or wifi: For replicating the network topology.

Plan Your SIP Simulation

Topology:
- SIP User Agents (UAs) interact through a directly (peer-to-peer) or SIP proxy server.
- For instance: Node A (UA1) ↔ Node B (Proxy) ↔ Node C (UA2).
Simulation Goals:
- To mimic SIP signaling such as session initiation, modification, and termination.
- We need to measure the performance indicators such as call configuration time, signaling overhead, and packet loss.

Create SIP Applications in NS3

While NS3 doesn’t include the SIP built-in, to utilize custom applications or external tools we want to execute it.

4.1 Steps to Create Custom SIP Applications

Define SIP Client and Server Classes:
- We can prolong the Application class, making SipClientApplication and SipServerApplication.
Simulate SIP Messages:
- We need to replicate the SIP commands such as INVITE, ACK, BYE, REGISTER, and so on, to utilize packet data.
Handle UDP/TCP Sockets:
- For SIP messages we can utilize UDP sockets and use TCP for session-related interaction.
State Management:
- We execute the state machines for session initiation, alteration, and termination.

4.2 Example: Basic SIP Client and Server Applications

Here’s a basic instance explaining the behaviour of SIP:

SIP Server

#include “ns3/core-module.h”

#include “ns3/network-module.h”

#include “ns3/internet-module.h”

#include “ns3/applications-module.h”

using namespace ns3;

class SipServer : public Application {

public:

SipServer();

virtual ~SipServer();

void Setup(Address address);

private:

virtual void StartApplication();

virtual void StopApplication();

void HandleRead(Ptr<Socket> socket);

Ptr<Socket> m_socket;

Address m_local;

};

SipServer::SipServer() : m_socket(0) {}

SipServer::~SipServer() {

m_socket = 0;

}

void SipServer::Setup(Address address) {

m_local = address;

}

void SipServer::StartApplication() {

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

m_socket->Bind(m_local);

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

}

void SipServer::StopApplication() {

if (m_socket) {

m_socket->Close();

m_socket = 0;

}

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

Ptr<Packet> packet;

Address from;

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

uint8_t* buffer = new uint8_t[packet->GetSize()];

packet->CopyData(buffer, packet->GetSize());

std::string data = std::string((char*)buffer, packet->GetSize());

NS_LOG_INFO(“SIP Server received: ” << data);

delete[] buffer;

// Simulate a 200 OK response

std::string response = “SIP/2.0 200 OK\r\n”;

Ptr<Packet> responsePacket = Create<Packet>((uint8_t*)response.c_str(), response.length());

socket->SendTo(responsePacket, 0, from);

}

SIP Client

class SipClient : public Application {

public:

SipClient();

virtual ~SipClient();

void Setup(Address address);

private:

virtual void StartApplication();

virtual void StopApplication();

void SendInvite();

void HandleRead(Ptr<Socket> socket);

Ptr<Socket> m_socket;

Address m_peer;

};

SipClient::SipClient() : m_socket(0) {}

SipClient::~SipClient() {

m_socket = 0;

}

void SipClient::Setup(Address address) {

m_peer = address;

}

void SipClient::StartApplication() {

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

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

Simulator::Schedule(Seconds(1.0), &SipClient::SendInvite, this);

}

void SipClient::StopApplication() {

if (m_socket) {

m_socket->Close();

m_socket = 0;

}

void SipClient::SendInvite() {

std::string invite = “INVITE sip:[email protected] SIP/2.0\r\n”;

Ptr<Packet> packet = Create<Packet>((uint8_t*)invite.c_str(), invite.length());

m_socket->SendTo(packet, 0, m_peer);

NS_LOG_INFO(“SIP Client sent: ” << invite);

}

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

Ptr<Packet> packet;

while ((packet = socket->Recv())) {

uint8_t* buffer = new uint8_t[packet->GetSize()];

packet->CopyData(buffer, packet->GetSize());

std::string data = std::string((char*)buffer, packet->GetSize());

NS_LOG_INFO(“SIP Client received: ” << data);

delete[] buffer;

}

Test and Debug

Enable Logging:

export NS_LOG=SipClient=level_all:SipServer=level_all

./waf –run scratch/sip-simulation

Inspect Packet Flows:
- Examine SIP messages within Wireshark to utilize PCAP tracing.

p2p.EnablePcapAll(“sip”);

Verify Protocol Logic:
- Confirm that INVITE, ACK, and other SIP messages are properly swapped.

Evaluate SIP Performance

Metrics to Measure:
- We want to compute the SIP performance indicators like call setup time (latency between INVITE and 200 OK), signaling overhead (bytes of control messages), and session duration and quality (optional).
Use FlowMonitor:

FlowMonitorHelper flowMonitor;

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

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

Advanced Features

Add Media Streams:
- After session initiation, we need to mimic media streams like RTP traffic.
Secure SIP (SIPS):
- For secure interaction, we execute SIP over TLS.
Proxy Servers:
- Insert SIP proxy servers transmitting requests among the clients.
Network Conditions:
- Launch delays, packet loss, or bandwidth limitations for robustness experimentation.

Document and Visualize

Document Implementation:
- Clearly define the SIP client/server logic, simulation metrics, and outcomes.
Visualize Results:
- Envision the interaction follow to utilize NetAnim tool.
- We need to graph the performance parameters with the help of Matplotlib or Gnuplot tools.

This module demonstrates how to setup and create SIP application then how to evaluate, visualize the performance outcomes using the simulation tool NS3. Further details will be provided about how the Session Initiation Protocol will perform in other simulation tool.

Our developers are engaged in refining the functionality of the Session Initiation Protocol (SIP), which serves as a signaling protocol for initiating, modifying, and terminating real-time communication sessions, including voice calls, video calls, and instant messaging. At phdprojects.org, we are committed to providing timely responses and delivering high-quality guidance for projects involving the Session Initiation Protocol using the NS3 tool. Please reach out to us to ensure the timely completion of your work.