To simulate Session Initiation Protocol (SIP) in ns2 has a series of steps to follow and it is a signalling protocol utilized for begin, maintaining, and end real-time sessions that includes voice, video, messaging, or other communications among devices on IP networks. SIP operates in combination with other protocols like Real-Time Transport Protocol (RTP) for actual media transport; however SIP itself is concentrated on session management and control.
Since NS2 does not directly support SIP, we can replicate SIP-like behaviour by designing the signalling and communication flows among nodes using TCP/UDP connections to replicate the control messages interchanged in the course of session establishment. We can also expand this with RTP traffic simulation for media streams once the SIP session is introduced.
The given below is a detailed procedures on how to simulate the Session Initiation Protocol in ns2.
Steps to Simulate SIP Projects Using NS2
- Install NS2
Ensure NS2 is installed on the system.
- Understanding SIP Traffic
In a typical SIP communication:
- User Agent Client (UAC): start the session (e.g., a SIP phone or client).
- User Agent Server (UAS): Responds to the session initiation request (such as a SIP server or another client).
- SIP INVITE: Transmitted by the UAC to request a session.
- SIP ACK: Acknowledgment from the UAS that the session is introduced.
- RTP Stream: Carries the actual voice/video data after the SIP signalling is done.
SIP messages are usually transmit over UDP, however we can also be carried over TCP for reliable transport, rely on the network environment.
- TCL Script for SIP-like Traffic Simulation
Below is a simple TCL script for replicating SIP signalling traffic using UDP connections among two nodes (one signifying the SIP client starts a call and the other signify the SIP server accepting it).
Example TCL Script for SIP Simulation
# Create a simulator object
set ns [new Simulator]
# Open files for tracing and NAM visualization
set tracefile [open sip_simulation.tr w]
$ns trace-all $tracefile
set namfile [open sip_simulation.nam w]
$ns namtrace-all $namfile
# Define network topology
set topo [new Topography]
$topo load_flatgrid 500 500
# Create two nodes: one as the SIP client and the other as the SIP server
set sip_client [$ns node]
set sip_server [$ns node]
# Define link parameters: bandwidth, delay, and queue type
set bw 10Mb
set delay 10ms
set queue DropTail
# Create a duplex link between the SIP client and the SIP server
$ns duplex-link $sip_client $sip_server $bw $delay $queue
# Attach a UDP agent to the SIP client node to simulate SIP signaling
set udp_client [new Agent/UDP]
$ns attach-agent $sip_client $udp_client
# Attach a UDP Sink agent to the SIP server node
set udp_server [new Agent/UDP]
$ns attach-agent $sip_server $udp_server
# Connect the SIP client (UDP) and SIP server (UDP)
$ns connect $udp_client $udp_server
# Create an Application/Traffic/CBR to simulate SIP INVITE, ACK, and BYE messages
set sip_traffic [new Application/Traffic/CBR]
$sip_traffic set packetSize_ 512 ;# Size of SIP messages (e.g., INVITE, ACK)
$sip_traffic set rate_ 64Kb ;# Simulating SIP signaling traffic
$sip_traffic attach-agent $udp_client
# Start SIP signaling (INVITE request) at 1.0 second and stop at 2.0 seconds
$ns at 1.0 “$sip_traffic start”
$ns at 2.0 “$sip_traffic stop”
# Simulate media session initiation (RTP) after SIP session is established
# Attach RTP UDP agent to both client and server
set rtp_client [new Agent/UDP]
$ns attach-agent $sip_client $rtp_client
set rtp_server [new Agent/UDP]
$ns attach-agent $sip_server $rtp_server
# Connect RTP session (after SIP signaling)
$ns connect $rtp_client $rtp_server
# Create RTP traffic over UDP (simulating voice or video data)
set rtp_traffic [new Application/Traffic/CBR]
$rtp_traffic set packetSize_ 1024 ;# Size of RTP packets (media traffic)
$rtp_traffic set rate_ 1Mb ;# Media traffic rate (1Mbps for video, 64kbps for voice)
$rtp_traffic attach-agent $rtp_client
# Start RTP traffic (media stream) at 3.0 seconds and stop it at 7.0 seconds
$ns at 3.0 “$rtp_traffic start”
$ns at 7.0 “$rtp_traffic stop”
# End simulation after 10 seconds
$ns at 10.0 “finish”
# Define the finish procedure to close trace and NAM files
proc finish {} {
global ns tracefile namfile
$ns flush-trace
close $tracefile
close $namfile
exec nam sip_simulation.nam &
exit 0
}
# Run the simulation
$ns run
- Explanation of the Script
- Simulator Setup: Generates a simulator object, trace, and NAM files to log events.
- Node Creation: Two nodes are generated— one denotes the SIP client starts the call, and the other denotes the SIP server (or another client) that accepts the call.
- UDP Connection: A UDP agent is attached to both the client and the server nodes to replicate SIP message interchanges over UDP.
- SIP Signaling: CBR traffic is utilized to replicate SIP signalling (such as INVITE, ACK, BYE messages). The packet size and rate are set to implement typical SIP message exchanges, with the SIP session initiation happening among 1.0 and 2.0 seconds.
- RTP Traffic: After the SIP signalling is done, RTP traffic is replicated using UDP to denotes the media (voice or video) being interchanged among the two nodes in the period of the session. The media stream initiates at 3.0 seconds and terminate at 7.0 seconds.
- Simulation End: The simulation terminates after 10 seconds, with SIP signalling first and media traffic following.
- Running the Simulation
Save the TCL script as sip_simulation.tcl, and execute it using NS2 with the following command:
ns sip_simulation.tcl
This will create a trace file (sip_simulation.tr) and a NAM file (sip_simulation.nam).
- Analysing the Results
- NAM Visualization: To envision the network and monitor the SIP signalling and RTP traffic among the client and server, open the NAM file using NAM (Network Animator):
nam sip_simulation.nam
NAM will demonstrate the interchange of SIP signalling messages over UDP (e.g., INVITE, ACK, BYE), followed by the media session denoted by RTP traffic.
- Trace File Analysis: The .tr file logs the events of the simulation. We need to evaluate this trace file to estimate the parameters like:
- SIP Message Delivery
- RTP Throughput
- End-to-End Delay
- Packet Loss
Example AWK Script to Calculate Throughput for RTP Traffic:
BEGIN { received_bytes = 0; start_time = 3; end_time = 7 }
{
if ($1 == “r” && $4 == “udp” && $8 == “rtp”) {
received_bytes += $5
}
}
END {
duration = end_time – start_time
throughput = (received_bytes * 8) / (duration * 1000000) # Convert to Mbps
print “RTP Throughput: “, throughput, “Mbps”
}
To utilize this AWK script, save it as rtp_throughput.awk and execute it on the trace file:
awk -f rtp_throughput.awk sip_simulation.tr
- Extending the Simulation
7.1. Multiple SIP Sessions
We can prolong the simulation to contain multiple SIP clients and servers, replicates multiple concurrent sessions:
# Add additional SIP clients and servers
set sip_client2 [$ns node]
set sip_server2 [$ns node]
# Create links and attach agents
$ns duplex-link $sip_client2 $sip_server2 $bw $delay $queue
set udp_client2 [new Agent/UDP]
$ns attach-agent $sip_client2 $udp_client2
set udp_server2 [new Agent/UDP]
$ns attach-agent $sip_server2 $udp_server2
# SIP session between client2 and server2
set sip_traffic2 [new Application/Traffic/CBR]
$sip_traffic2 set packetSize_ 512
$sip_traffic2 set rate_ 64Kb
$sip_traffic2 attach-agent $udp_client2
$ns at 1.5 “$sip_traffic2 start”
$ns at 2.5 “$sip_traffic2 stop”
7.2. Simulating Call Failures
We can replicate SIP session failures, like packet loss in signalling, by incorporating a packet loss model:
# Add packet loss to simulate call failures
$ns lossmodel [new ErrorModel/Uniform] 0.05 $sip_client $sip_server
7.3. Varying Network Conditions
Adjust the link bandwidth, delay, or incorporate jitter to replicate changing network conditions, and monitor their impacts on SIP and RTP traffic.
We had successfully provide the comprehensive guide to simulate SIP-like behaviour using NS2 by design the signalling flows for session initiation, and optionally, incorporate the media traffic after session establishment. More information will be shared in the upcoming manual.
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