A simulation-based comparative evaluation of transport protocols for SIP Authors: M.Lulling*, J.Vaughan Department of Computer science, University college.

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Presentation transcript:

A simulation-based comparative evaluation of transport protocols for SIP Authors: M.Lulling*, J.Vaughan Department of Computer science, University college Cork, Western Road, Cork, Ireland. Publication: ELSEVIER on Computer communications, April 2005 Reporter: Chun-Hui Sung Date: 2007/5/24

Outline Introduction Transport for SIP Simulations Results Conclusion Comment

Introduction Uses the Network Simulator – NS2 to investigate the direct effects and subsequent consequences associated with the use of different transport protocols in a SIP context. Performance evaluation in the result of VoIP SIP signaling from simulation-based experiments underlying transport protocol.

Introduction ( Cont. ) SIP (Session Initiation Protocol) is A peer-to-peer protocol An application layer signaling protocol Create, modify and terminate sessions Applications can be voice, video, gaming, instant messaging, presence, call control, etc.

Transport for SIP SIP over TCP TCP Reno TCP Vegas TCP Sack SIP over UDP SIP over SCTP

SIP over TCP TCP - Reno

SIP over TCP TCP - Vegas

SIP over TCP TCP - Sack

SIP over UDP The user datagram protocol, UDP, is a connectionless transport protocol that does not provide any guarantee of message delivery.

SIP over SCTP The stream control transmission protocol, SCTP, is a reliable end-to-end transport layer protocol, and while support for TCP and UDP is included in the core SIP specifiation.

Simulations Network topology Node 1 and 2 are buffer-limited droptail routers, all other nodes are endpoints, node 1 and 2 are the only bottleneck link. Node 0 and 3 are SIP proxies. Node 4 and 5 are used to provide competing cross-traffic. Router SIP Proxy ASIP Proxy B

Simulations ( Cont. ) NS2 parameters: Delay time are 45 ms between the proxies. The simulations use a stationary Poisson model to generate the arrival times of 512-byte session establishment requests at node 0. Individual SIP requests are independent and are generated at node 0 at 160/s, which corresponds to a link utilization of approximately 33% on the bottleneck link.

Simulations ( Cont. ) Induced packet loss Random packet loss Competing traffic Throughput analysis

Simulations ( Cont. ) Induced packet loss In order to measure and evaluate the delays and delaying effects of packet loss on the system, packets are explicitly dropped from node 1. The simulation is run 10 times for each of the five transport protocols or variants. The time at which each message is generated by the application at node 0 and the time at which this message is passed to the application at node 3 is recorded. (delay time)

Simulations ( Cont. ) Random packet loss Random packet loss percentages of between 0.1 and 0.5% (in 0.1% intervals) are simulated at node 1 with uniform distribution. The time at which each message is generated by the application at node 0 and the time at which this message is passed to the application at node 3 is recorded. (delay time)

Simulations ( Cont. ) Competing traffic Simulate the effects of cross-traffic generated between node 4 and 5, providing competition for bandwidth on the bottleneck link between nodes 1 and 2. TCP Reno is used exclusively as the transport protocol for the competing traffic in all simulations. Delays are measured as describe in the two previous experiments.

Simulations ( Cont. ) Throughput analysis Add a variable of buffer size at node 1 The simulations have been run with buffer sizes of 5, 20, 50, 100, 150, 200 and 250 packets at node 1. The default value of buffer size is 50.

Results Induced packet loss Random packet loss Competing traffic Throughputs

Results – Induced packet loss (1/3) [ Five consecutively dropped packets ]

Results – Induced packet loss (2/3) [ Peak delays ]

Results – Induced packet loss (3/3) [ Message affects ]

Results – Random packet loss (1/4) [ loss rate of 0.3 % ] SIP traffic with TCP Reno SIP traffic with SCTP

Results – Random packet loss (2/4) [loss rate of 0.3% ] SIP traffic with TCP Sack SIP traffic with UDP

Results – Random packet loss (3/4) [loss rate of 0.3% ] SIP traffic with TCP Vegas

Results – Random packet loss (4/4) [ loss rate of 0.1% ~ 0.5% ] Mean delay per packet loss percentage

Results – competing traffic (1/4) [ SIP traffic with TCP Reno ]

Results – competing traffic (2/4) [ SIP traffic with UDP / SCTP / TCP SACK ]

Results – competing traffic (3/4) [ % overall throughput ]

Results – competing traffic (4/4) [ Total throughput ]

Results – Throughputs Mean throughput for SIP vs. FTP TCP Sack Mean throughput for SIP vs. FTP TCP Vegas

Conclusion Authors compare and analyze the performance of SIP over UDP [TCP-Reno/Vegas/Sack, SCTP]. This paper was found that TCP Sack and SCTP are the best options for a reliable transport protocol for SIP traffic.

Comment They don’t put attention on multi-homing of SCTP.