How to Start Real Time Protocol Projects Using NS3

To start a Real-Time Protocol (RTP) project in NS3, we need to replicate the transmission of real-time audio, video, or other data streams through network. Since NS3 doesn’t directly support a built-in RTP execution then we can be replicated the behavior of RTP to utilise custom applications because RTP is normally taken over UDP.

Given below is a stepwise method to start Real-Time Protocol Projects in NS3:

Steps to Start Real Time Protocol Projects in NS3

  1. Understand RTP and Its Simulation Requirements
  • What is RTP?
    • RTP is described within RFC 3550 for distributing the real-time data such as audio, video.
    • It functions over UDP for lightweight delivery including optional RTP Control Protocol (RTCP) for quality monitoring.
    • Contains:
      • Sequence Numbers: Identifying the lost packets.
      • Timestamps: For synchronization, it utilizes.
      • Payload Type: Detecting data formats.
  • Key Aspects to Simulate:
    • Packet generation simulating the RTP streams.
    • Real-time delivery including sequence numbers and timestamps.
    • Performance parameters like latency, jitter, and packet loss.
  1. 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 UDP/IP interaction.
    • applications: For custom or bulk data applications.
    • point-to-point or wifi: These modules used for replicating the network topology.
  1. Plan Your RTP Simulation
  • Topology:
    • A basic sender-receiver configuration or a multicast scenario:
      • Node A (RTP sender) → Node B (RTP receiver).
      • Node A (RTP sender) → Multiple receivers through multicast.
  • Simulation Goals:
    • To mimic RTP packet generation, delivery, and monitoring.
    • We estimate the performance indicators such as end-to-end delay, jitter, and packet loss.
  1. Create a Custom RTP Application

While NS3 doesn’t contain RTP built-in then we want to execute it to utilize custom applications.

4.1 Steps to Create a Custom RTP Application

  1. Define RTP Sender and Receiver Classes:
    • We can prolong the Application class making RtpSenderApplication and RtpReceiverApplication.
  2. Simulate RTP Packet Format:
    • It contains RTP headers such as sequence number, timestamp, payload type.
    • Sum up RTP headers and payload to utilize NS3’s Packet class.
  3. Handle UDP Communication:
    • For RTP data transmission, we need to utilize UDP sockets.
  4. Monitor Performance:
    • Insert logic, estimating the packet loss, jitter, and latency.

4.2 Example: Basic RTP Sender and Receiver

RTP 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 RtpSender : public Application {

public:

RtpSender();

virtual ~RtpSender();

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;

uint32_t    m_sequenceNumber;

double      m_interval; // Time interval between packets

};

RtpSender::RtpSender() : m_socket(0), m_sequenceNumber(0) {}

RtpSender::~RtpSender() {

m_socket = 0;

}

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

m_peerAddress = address;

m_payloadSize = payloadSize;

m_interval = interval;

}

void RtpSender::StartApplication() {

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

m_socket->Connect(m_peerAddress);

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

}

void RtpSender::StopApplication() {

if (m_socket) {

m_socket->Close();

m_socket = 0;

}

Simulator::Cancel(m_sendEvent);

}

void RtpSender::SendPacket() {

// Simulate RTP header

uint8_t rtpHeader[12];

memset(rtpHeader, 0, sizeof(rtpHeader));

rtpHeader[0] = 0x80; // Version 2

rtpHeader[1] = 96;   // Payload type

rtpHeader[2] = (m_sequenceNumber >> 8) & 0xFF;

rtpHeader[3] = m_sequenceNumber & 0xFF;

m_sequenceNumber++;

// Create payload

uint8_t* payload = new uint8_t[m_payloadSize];

memset(payload, 0xAB, m_payloadSize); // Example payload

// Combine RTP header and payload into a packet

Ptr<Packet> packet = Create<Packet>(payload, m_payloadSize);

packet->AddHeader(rtpHeader, sizeof(rtpHeader));

m_socket->Send(packet);

delete[] payload;

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

}

RTP Receiver

class RtpReceiver : public Application {

public:

RtpReceiver();

virtual ~RtpReceiver();

void Setup(Address address);

private:

virtual void StartApplication();

virtual void StopApplication();

void HandleRead(Ptr<Socket> socket);

Ptr<Socket> m_socket;

Address     m_local;

};

RtpReceiver::RtpReceiver() : m_socket(0) {}

RtpReceiver::~RtpReceiver() {

m_socket = 0;

}

void RtpReceiver::Setup(Address address) {

m_local = address;

}

void RtpReceiver::StartApplication() {

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

m_socket->Bind(m_local);

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

}

void RtpReceiver::StopApplication() {

if (m_socket) {

m_socket->Close();

m_socket = 0;

}

}

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

Ptr<Packet> packet;

Address from;

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

uint8_t rtpHeader[12];

packet->CopyData(rtpHeader, 12);

uint32_t sequenceNumber = (rtpHeader[2] << 8) | rtpHeader[3];

NS_LOG_INFO(“Received RTP packet with sequence number: ” << sequenceNumber);

}

}

  1. Test and Debug
  • Enable Logging:

export NS_LOG=RtpSender=level_all:RtpReceiver=level_all

./waf –run scratch/rtp-simulation

  • Inspect Packet Flows:
    • Examine the RTP packets within Wireshark to utilize PCAP tracing.

p2p.EnablePcapAll(“rtp”);

  1. Evaluate RTP Performance
  • Metrics to Measure:
    • Latency: Compute the duration for packets attaining the receiver.
    • Jitter: Variability within packet arrival times.
    • Packet Loss: We measure the rate of packets that are lost.
  • Use FlowMonitor:

FlowMonitorHelper flowMonitor;

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

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

  1. Advanced Features
  • Multicast RTP:
    • We execute the multicast transmission to numerous receivers to utilize NS3’s multicast support.
  • RTCP Integration:
    • Insert RTCP for quality monitoring and reporting.
  • Simulate Media Streams:
    • Make audio or video-like payloads including diverse data rates.
  • Simulate Network Conditions:
    • We need to launch packet loss, delays, and jitter estimate the robustness of RTP.

This project allows you to simulate the Real Time Protocol projects and to evaluate the performance using offered methodology within NS3 tool. More information will be shared regarding these projects in another manual.

Our team of developers is ready to enhance your project’s performance, so don’t hesitate to contact us for additional help. If you’re starting a Real Time Protocol project using NS3, you can rely on the experts at phdprojects.org to assist you at every stage. Make sure to utilize the personalized support from phdprojects.org to get the best outcomes and meet your deadlines.