Evaluation of UDP and SCTP for SIP-T and TCP, UDP and SCTP with constant traffic

Sammanfattning: In recent years, Voice over IP (VoIP) has gained a lot of popularity. Signaling being an important part of VoIP has been addressed by the (IETF) SIGTRAN working group to meet Quality of Service as given by Public Switched Telephone Network (PSTN), so that both PSTN and VoIP can co-exit and work together in a seamless manner SIP (Session Initiation Protocol) developed by IETF for VoIP signaling is a communication control protocol capable of running on different transport layers, e.g., TCP, UDP or SCTP. Today’s SIP application is mostly operating over the unreliable transport protocol UDP. In lossy environment such as wireless networks and congested Internet networks, SIP messages can be lost or delivered out of sequence. The SIP application then has to retransmit the lost messages and re-order the received packets. This additional processing overhead can degrade the performance of the SIP application. Therefore to solve this problem, the researchers are looking for a more appropriate transport layer for SIP. SCTP, a transport protocol providing acknowledged, error-free, non-duplicated transfer of messages, has been proposed to be an alternative to UDP [1] and TCP [2]. The multi-streaming and multi-homing features of SCTP are especially attractive for applications that have stringent performance and high reliability requirements and an example is the SIP proxy server. In this research, we have analyzed the performance offered by SCTP for SIP message delivery in the perspective of historic research work as well as determined call setup time using UDP and SCTP by simulating SIP traffic in Network Simulator-2 (ns-2). We also evaluate TCP, UDP and SCTP traffic with constant bit rate of traffic through ns-2