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A simulation-based comparative evaluation of transport protocols for SIP

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Uses the Network Simulator NS2 to investigate the direct effects and ... SIP over SCTP ... Delay time are 45 ms between the proxies. ... – PowerPoint PPT presentation

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Title: A simulation-based comparative evaluation of transport protocols for SIP


1
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

2
Outline
  • Introduction
  • Transport for SIP
  • Simulations
  • Results
  • Conclusion
  • Comment

3
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.

4
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.

5
Transport for SIP
  • SIP over TCP
  • TCP Reno
  • TCP Vegas
  • TCP Sack
  • SIP over UDP
  • SIP over SCTP

6
SIP over TCP
  • TCP - Reno

7
SIP over TCP
  • TCP - Vegas

8
SIP over TCP
  • TCP - Sack

9
SIP over UDP
  • The user datagram protocol, UDP, is a
    connectionless transport protocol that does not
    provide any guarantee of message delivery.

10
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.

11
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.

12
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.

13
Simulations ( Cont. )
  • Induced packet loss
  • Random packet loss
  • Competing traffic
  • Throughput analysis

14
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)

15
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)

16
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.

17
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.

18
Results
  • Induced packet loss
  • Random packet loss
  • Competing traffic
  • Throughputs

19
Results Induced packet loss (1/3) Five
consecutively dropped packets
20
Results Induced packet loss (2/3) Peak
delays
21
Results Induced packet loss (3/3) Message
affects
22
Results Random packet loss (1/4) loss rate of
0.3
  • SIP traffic with TCP Reno
  • SIP traffic with SCTP

23
Results Random packet loss (2/4) loss rate of
0.3
  • SIP traffic with TCP Sack
  • SIP traffic with UDP

24
Results Random packet loss (3/4) loss rate of
0.3
  • SIP traffic with TCP Vegas

25
Results Random packet loss (4/4) loss rate
of 0.1 0.5
  • Mean delay per packet loss percentage

26
Results competing traffic (1/4) SIP traffic
with TCP Reno
27
Results competing traffic (2/4) SIP traffic
with UDP / SCTP / TCP SACK
28
Results competing traffic (3/4) overall
throughput
29
Results competing traffic (4/4) Total
throughput
30
Results Throughputs
  • Mean throughput for SIP vs. FTP TCP Sack
  • Mean throughput for SIP vs. FTP TCP Vegas

31
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.

32
Comment
  • They dont put attention on multi-homing of SCTP.
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