How to Implement Voice over IP Protocol in NS2

To implement Voice over IP (VoIP) in NS2 has needs to mimic the transmission of voice data over IP networks using protocols such as RTP for media delivery and SIP or H.323 for call setup and signalling. In NS2, VoIP can be mimics by generating traffic patterns that denotes the features of voice communication, like small, frequent packet transmissions over UDP. The following are the brief implementation steps to execute the VoIP in ns2:

Steps to Implement VoIP in NS2

  1. Set Up NS2 Environment
    • Make sure that NS2 is installed and setup on the system.
  2. Understand VoIP Protocol Stack
    • RTP (Real-Time Protocol): Used for delivering voice data.
    • UDP: Used as the transport layer for RTP because of its low-latency features.
    • SIP or H.323: It is used for call setup and signalling. Since SIP or H.323 emulate be complex, RTP can be used to signify the actual voice data transmission.
  3. Simulate VoIP Traffic Using RTP over UDP
    • Generate a custom application in OTcl that creates RTP-like traffic over a UDP connection. This mimics the transmission of voice packets in a usual VoIP scenario.
  4. Create a Basic VoIP Simulation in NS2

The below are the instances of OTcl script that mimic the VoIP traffic among two nodes:

# Define the simulation environment

set ns [new Simulator]

# Open the trace file

set tracefile [open out.tr w]

$ns trace-all $tracefile

# Define the finish procedure

proc finish {} {

global ns tracefile

$ns flush-trace

close $tracefile

exec nam out.nam &

exit 0

}

# Create VoIP sender and receiver nodes

set voip_sender [$ns node]

set voip_receiver [$ns node]

# Create a duplex link between sender and receiver

$ns duplex-link $voip_sender $voip_receiver 10Mb 10ms DropTail

# Create UDP agent for VoIP sender and attach it to the node

set udp_sender [new Agent/UDP]

$ns attach-agent $voip_sender $udp_sender

# Create UDP agent for VoIP receiver and attach it to the node

set udp_receiver [new Agent/UDP]

$ns attach-agent $voip_receiver $udp_receiver

# Connect UDP agents

$ns connect $udp_sender $udp_receiver

# Create a custom application to simulate VoIP traffic

set voip_app [new Application/Traffic/Pareto]

$voip_app attach-agent $udp_sender

$voip_app set packetSize_ 160        ;# Typical size of an RTP voice packet (e.g., G.711 codec)

$voip_app set burst_time_ 0.02       ;# Packetization interval (20ms)

$voip_app set idle_time_ 0.02        ;# Continuous stream with minimal idle time

$voip_app set rate_ 64Kb             ;# Bandwidth rate for VoIP stream

# Start the VoIP traffic at 1.0s

$ns at 1.0 “$voip_app start”

# Stop the simulation at 10.0s

$ns at 10.0 “finish”

# Run the simulation

$ns run

  1. Explanation of the Script
    • Simulator Object ($ns): Configure the simulation environment.
    • Nodes: Two nodes are generated to denote the VoIP sender and receiver.
    • UDP Agents: Since RTP usually executes over UDP, UDP agents are attached to the sender and receiver nodes.
    • Custom VoIP Application: A custom application using Traffic/Pareto is generated to mimic the VoIP traffic. This model is appropriate for the bursty, real-time nature of voice traffic.
    • Packet Size and Timing: The packet size and timing intervals are properly to mimic the usual VoIP streams, with small packets communicated every 20ms, simulate a codec such as G.711.
  2. Run the Simulation
    • Save the OTcl script to a file like voip_simulation.tcl.
    • Execute the script using NS2:

ns voip_simulation.tcl

    • The simulation will create a trace file (out.tr) and a Network Animator (NAM) file (out.nam). We can visualize the simulation using NAM.
  2. Analyse the Results
    • The trace file (out.tr) will include elaborate information about the packets sent, received, and dropped. We can evaluate the performance of the VoIP traffic that has parameters such as latency, jitter, packet loss, and throughput.
  3. Enhancements
    • Multiple VoIP Streams: To mimic multiple VoIP calls by generating additional VoIP sender-receiver pairs.
    • QoS Mechanisms: Execute Quality of Service (QoS) approaches like DiffServ or MPLS to select VoIP traffic and minimize latency and jitter.
    • Network Impairments: Establish packet loss, jitter, and latency to mimic real-world network conditions and monitor their impacts on VoIP quality.
  4. Additional Considerations
    • Codec Simulation: Adjust the packet size and interval based on diverse voice codecs like G.729 for lower bandwidth usage.
    • Call Setup Simulation: Since the emphasis here is on media delivery, we can mimic SIP or H.323 signalling if desired, nevertheless it would essential more complex scripting.
    • Security: To mimic Secure RTP (SRTP) by establishing encryption mechanisms, while this contains more advanced modifications to the simulation.

In this manual, we get knowledge about how to setup the simulation and how to execute the Voice over IP protocol in the ns2 simulator. More information will be shared about the Voice over IP protocol and its execution in different simulations.

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