LOGO 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
LOGO 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
LOGO SIP Overview End – to – end design Intelligent resides in end device Network maintain almost zero intelligent Result: Flexibility Scalability
LOGO 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
LOGO 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
LOGO 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
LOGO 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.
LOGO 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
LOGO 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  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
LOGO 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.
LOGO 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
LOGO 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.
LOGO 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
LOGO 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
LOGO 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
LOGO 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 , 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.
LOGO 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.
LOGO 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.
LOGO 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 . 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.
LOGO 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 . As mentioned before, these delays cause considerable packet loss, which adversely affects the QoS of multimedia streaming applications. 
LOGO References  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.  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.  R. Mahy and D. Petrie, “The Session Initiation Protocol (SIP) “Join” Header”, draft-ietf-sip-join- 03.txt, Feb 2004, Work in progress.  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.  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.  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.  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.  T. Eyers and H. Schulzrinne, “Predicting Internet Tele-phony Call Setup Delay”, Proceedings of 1st IP-Telephony Wksp., Berlin, Germany, Apr. 2000.  L. Kleinrock, QUEUING SYSTEMS Volume I: Theory, John Wiley & Sons, 1975.  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.