1. SIP Over Wireless Network
Harry Sunarsa

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Outline
1 Introduction
2 Brief SIP Overview
3 SIP Mobility Support
4 Handoff Delay Analysis

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2 SIP Overview

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SIP Overview
 Real-time communication protocol for VOIP-
Voice over IP and has been expanded to
support video and instant-messaging
applications.
 Designed to perform basic call-control tasks,
such as session call set-up and signaling
 Text-based protocol
 However, individual functions are served by
separate protocols. i.e. Media transport uses
RTP/RTCP

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SIP Overview
Network Element
 User Agent: UA Client & UA
Server
 Server.. (Logical server)
• Registrar: Maintain user’s
whereabouts at a location
server
• Proxy: Relay call signaling
• Redirect: Redirect user to other
server
 Gateway

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SIP Overview

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SIP Overview
End – to – end design
Intelligent resides in end device
Network maintain almost zero intelligent
Result:
 Flexibility
 Scalability

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SIP Overview
 The message types are typically Request-Response
messages, either a request from a client to a server, or a
response from a server to a client

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SIP Overview
 Request methods
Message Name Function
REGISTER Register a user with a SIP server (with
location service)
INVITE Invite user(s) to a session. The body of
the Message contains the description
with the address where the host wants
to receive the media stream
ACK Acknowledgement of an INVITE request
CANCEL Cancel a pending request
BYE Terminate a session (release a call)
OPTION Query servers about their capabilities

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SIP Overview
 Respond methods
Code Response type Description
Classes
1xx Provisional Request received, continuing to process
the request
2xx Success The action was successfully received,
understood and
accepted
3xx Redirection Further action needs to be taken in order
to complete the
request
4xx Client Error The request contains bad syntax or cannot
be fulfilled at this
server
5xx Server Error The server failed to fulfill an apparently
valid request
6xx Global Failure The request cannot be fulfilled at any
server

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SIP Overview
 Sample of SIP interaction

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SIP Overview
 Packetized Communication

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3 SIP Mobility Support

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Mid-call Mobility
 the terminal needs to intimate the correspondent host (CH) or the host
communicating with the MH, by sending an INVITE message about the
terminal’s new IP address and updated session description. Hence, the
handoff delay is essentially the one-way delay for sending an INVITE
message from the MH to the CH.

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Handoff Procedure
 Each base station is equipped with a SIP B2BUA and a SIP proxy server
 A B2BUA is a logical entity that receives a request and processes it as a
user agent server (UAS)
 It maintains dialog state and participates in all requests sent on the
dialogs it has established
 All the SIP messages are directed through the outbound proxy at the
base station using the Record-Route field of the message header, so that
the B2BUA is able to capture the ongoing dialog information
 The B2BUA is coupled with a media gateway that acts as a proxy,
forwarding the RTP packets
 The media gateway has two functions, such as: as RTP packet replicator
and as RTP packet filter

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Handoff Procedure
Sending of JOIN message to initiate the soft handoff
 During the transition period when a new network interface gets activated,
the SIP UAC at the MH sends an INVITE message with the JOIN header
[5] to the SIP B2BUA proxy server
 The JOIN header contains all the relevant information about the ongoing
call
 The B2BUA essentially, configures the packet replicator at the media
gateway to send a copy of all packets directed towards the old interface
of the MH to the newly activated interface

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Handoff Procedure
Splitting of RTP stream – soft handoff procedure
 During the transient handoff period the MH sends and receives
the packets through both the interfaces.

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Handoff Procedure
signaling to update ongoing session parameters on account of the change in MH’s IP
address
 The packet filters at the media gateway and the MH discards the
duplicate RTP packets. As soon as the packet reaches the MH
through the newly activated interface, a reINVITE message is
sent to the CH with the new IP address and corresponding
contact information

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Handoff Procedure
Message Diagram

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Handoff Procedure

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4 Handoff Delay Analysis [9]

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Handoff Delay Analysis
 According to SIP the mid-call procedure described in the
previous section the handoff delay is essentially the one-way
transmission time of the INVITE message from MH to the CH and
its subsequent processing time at the CH.
 hand-off delay is typically the time required for the INVITE
message from the MH to reach the CH. Major delays in this
hand-off procedure occur at (i) the MH, (ii) the wireless radio link
between the MH and the BS, (iii) the Internet, and (iv) the CH or
the server.

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Handoff Delay Analysis
 Here is assumed that M/M/1 queuing model is used at the MH
and the base station, and a priority based M/G/1 model for the
CH or the server

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Handoff Delay Analysis
 The hand-off delay (Dhandoff) in transmitting a SIP message can be
computed as figure:
 Where
D1= delay at the MH,
D2= the delay incurred in transmitting the SIP message over the
wireless link,
D3=the queuing delay in the BS,
D4= the constant of internet transmission delay,
D5= Queuing delay in the CH

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Handoff Delay Analysis
 Where
X1, X2 are the second moment of μ1 and μ2 respectively.
 To derive D2, require to adopt a delay model over wireless links
 TCP being an end-to-end protocol, its error recovery mechanism
is not appropriate for real-time transmission
 This is because end-to-end retransmission is not recommended
for real-time applications to avoid delay variance

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Handoff Delay Analysis
 semi-reliable link layer retransmission mechanism like Radio Link
Protocol (RLP) is used to reduce the air link FER and thus
increase reliability. RLP works on the basis of a NAK based
acknowledgment scheme.
 According to the model used and the results reported in [12], the
delay to transmit a TCP segment consisting of k frames over a
radio link without RLP operating on it, is given by
 is the packet loss rate, while p is the probability of a
frame being in error in the air link.
 Nm is the number of TCP re-transmission before a successful
transmission, τ is the inter-frame time. D is the end-to-end frame
propagation delay over the radio channel.
 The typical value are D= 100 msec and τ =20msec.

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Handoff Delay Analysis
 When the RLP is used in order to reduce the re-transmission, the
term will be:
 Where n=3 is the maximum number of RLP retransmission trials.
 Ci,j the first frame received correctly at the destination, is the ith
retransmission frame at the jth retransmission trial.
 To derive the value of k, assumed that a TCP segment is carried in one
packet (note that frames here imply air link frames). Assume that the air
link frame duration is 20 msec. Therefore, a 9.6Kbps radio channel
contains 9.6 x 103 x 10 -3 x 20 x 1/8 =24 bytes in each frame. Also it is
assumed that the size of one SIP message is 500 bytes. Therefore
number of air link frames in a SIP message is 500/24=21.

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Handoff Delay Analysis
Comparison of delay in a 9.6 Kbps channel Comparison of delay in a 9.6 Kbps channel with FER=
with arrival rate 500 messages/sec of SIP 0.05
messages at the BS
Typically, when RLP is used, the delay in transmitting the SIP message has
been found to be 6.42 secs for FER = 0.5. Here also the delay incurred in
the wireless part has been found to be 98% of the total end-to-end delay.
The delay reduces with the use of RLP, which takes care of the longer TCP
retransmissions.

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Handoff Delay Analysis
 Results show that the major portion (more than) of the handoff
delay is due to the wireless link.
 the delay for moderate FER with RLP is found to be around 6
seconds [9]. But media streams can function normally with a
maximum interruption of 50 msec, while an interruption of 200
msec is generally acceptable and an interruption of 500 msec
causes a perceptible real interruption, which is unacceptable.
 Hence the SIP based mobility management is not suitable for
media streaming in wireless networks.
 Even if application layer solution for supporting mobility may seem
to be an attractive option, it is not quite suitable for delay-sensitive
media streaming.

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Summary
 The handoff procedure SIP based is composed of the following
major operations, each of which contributes to the handoff delay:
 (i) Network detection and address configuration operation performed
by the MH. It depends on the networking technology and the MH’s
operating system.
 (ii) Sending the INVITE message with the JOIN header to BS_I.
 (iii) Sending the re-INVITE message to update the session with the
new location parameters.
 SIP provides an elegant application layer mobility support that
solves the problems associated with lower layer mobility protocols
in next generation heterogeneous wireless access networks.
However, the handoff delay in SIP may be substantial causing
considerable packet loss, which affects the quality of voice or video
streams seriously [3].
 As mentioned before, these delays cause considerable packet
loss, which adversely affects the QoS of multimedia streaming
applications. [3]

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References
[1] J.Kuthan and D. Sisalem, “Tutorial SIP: More Than You Ever Wanted To Know About”, TEKELEC.
*2+ “Capitolo 1: Il Protocollo SIP”, Università degli Studi di Napoli Fedrico II, Facolta di Ingeneria.
[3] N. Banerjee, S.K. Das and A. Acharya “SIP-based Mobility Architecture for Next Generation
Wireless Networks”, Proceedings of the 3rd IEEE Int’l Conf. on Pervasive Computing and
Communications (PerCom 2005).
*4+ M. Handley and V. Jacobson, “SDP: Session Description Protocol”, RFC 2327, April 1998.
[5] R. Mahy and D. Petrie, “The Session Initiation Protocol (SIP) “Join” Header”, draft-ietf-sip-join-
03.txt, Feb 2004, Work in progress.
[6] E. Wedlund and H. Schulzrinne, “Mobility Support using SIP”, Proceedings of ACM/IEEE
International Workshop on Wireless and Mobile Multimedia (WoWMoM), Page(s): 76-82, August
1999.
[7] N. Banerjee, W. Wu, S.K. Das, S. Dawkins, J. Pathak, “Mobility support in wireless Internet”, IEEE
Wireless Communications 10 (5) (2003) 51–61. October, 2003.
[8] N. Banerjee, W. Wu, S.K. Das, K. Basu, “Analysis of SIP-based mobility management in 4G
wireless networks”, Center for Research in Wireless Mobility and Networking (CReWMaN),
Department of Computer Science and Engineering, The University of Texas at Arlington, Arlington.
[9] N. Banerjee, S.K. Das, K. Basu, “Hand-off Delay Analysis in SIP-based Mobility Management in
Wireless Networks”, Center for Research in Wireless Mobility and Networking (CReWMaN),
Department of Computer Science and Engineering, The University of Texas at Arlington, Arlington.
[10] T. Eyers and H. Schulzrinne, “Predicting Internet Tele-phony Call Setup Delay”, Proceedings of
1st IP-Telephony Wksp., Berlin, Germany, Apr. 2000.
[11] L. Kleinrock, QUEUING SYSTEMS Volume I: Theory, John Wiley & Sons, 1975.
[12] S. K. Das, E. Lee, K. Basu, N. Kakani, and S.Sen, “Performance Optimization of VoIP Calls over
Wireless Links Using H.323 Protocol”, Proceedings of the 2002 INFOCOM, Pages: 1386-1394, 2002.

32. If you never try, you’ll never know