Session Initiation Protocol (SIP) The Emerging System in IP Telephony

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1 Session Initiation Protocol (SIP) The Emerging System in IP Telephony

2 Introduction Session Initiation Protocol (SIP) is an application layer control protocol that can establish, modify and terminate multimedia sessions or calls. The protocol is developed by IETF in their series of proposals for the provision of advanced telephony services across the Internet. The latest version in SIP being Version 2 proposed in March The chief applications that can be supported by SIP include multimedia conferences, distance learning and Internet telephony and multimedia distribution. The strength of SIP lies in it s simplicity, scalability, extensibility, and modularity. SIP can invite both persons and robots like media storage services. SIP can invite parties to both unicast and multicast sessions. SIP can be used not only to initiate sessions but also to sessions that have been advertised and established by other means, such as, Session Announcement Protocol (SAP), electronic mail, news groups, web pages or directories (LDAP) etc., SIP transparently supports name mapping and redirection services, allowing the implementation of ISDN and Intelligent Network telephony subscriber services, which also facilitates personal mobility. SIP supports the following features in establishing and terminating multimedia communications. User Location: SIP determines the end system to be used for communication. User Capabilities: SIP negotiates the type of media and media parameters to be used for communication. User Availability: SIP determines the willingness of the called party to engage in communications. Call Set up and Handling: SIP establishes, maintains and terminates the call. SIP also supports other protocols like RSVP (Resource Reservation Protocol) for reserving network resources, RTP (Real Time Protocol) and RTCP (Real Time Control Protocol) for transporting real-time data and providing QoS feedback respectively, RTSP (Real Time Streaming Protocol) for controlling delivery of streaming media, SAP (Session Announcement Protocol) for advertising multimedia sessions Via multicast and SDP (Session Description Protocol) for describing multimedia sessions. However the functionality and operation of SIP does not depend on any of these protocols. SIP can also be used in conjunction with other call setup and signaling protocols. For example, SIP could be used to determine that the party can be reached Via H.323, obtain the H.245 gateway and user address and then use H to establish a call.

3 2.0 SIP features at a glance SIP can establish, modify and terminate multimedia calls and sessions and can invite both persons and robots like media storage service. SIP is a text-based protocol, which uses ISO in UTF-8 encoding, thereby makes it flexible and extensible. SIP locates user given -style address. SIP transparently supports name mapping and redirection services, allowing the implementation of ISDN and Intelligent Network telephony subscriber services. SIP can (Re-) negotiate session parameters using SDP (Session description protocol) during the call initiation and during the call respectively. SIP supports manual and automatic forwarding (name/number mapping). SIP is network protocol independent. It can work on both TCP and UDP. SIP Proxy server can fork the INVITE call to the multiple addresses, as returned by the Registration Server, thereby reducing the call set up time. SIP handles termination and transferring of calls. In case of SIP on the top of UDP, retransmission scheme is used to enhance the reliability of the protocol. SIP provides call control (hold, forward, transfer, media changes, ) SIP can handle both unicast and multicast sessions SIP can also initiate multi-party calls using a multipoint control unit (MCU) or fully meshed interconnection instead of multicast. Internet telephony gateways that connect Public Switched Telephone Network (PSTN) parties can also use SIP to set up calls between them. SIP can invite users to sessions with or without resource reservation. SIP do not reserve resources, but can convey to the invited system, the information necessary to do this.

4 2.1 Typical components in a SIP system The typical components in a SIP system are user agent server, user agent client, proxy server, redirect server and registrar or location server. They are briefly described below. User agent client (UAC): calling user agent: A user agent client is a Client application that initiates the SIP request. User agent server (UAS): called user agent: A user agent server is a server application that contacts the user when a SIP request is received and that returns a response on behalf of the user. The response accepts, rejects or directs the request. Proxy, proxy server: An intermediary program that acts as both a server and a client for the purpose of making requests on behalf of other clients. Requests are serviced internally or by passing them on, possibly after translation, to other servers. A proxy interprets, and, if necessary, rewrites a request message before forwarding it. Redirect server: A redirect server is a server that accepts a SIP request, maps the address into zero or more new addresses and returns these addresses to the client. Unlike a proxy server, it does not initiate its own SIP request. Unlike a user agent server, it does not accept calls. Registrar: A registrar is a server that accepts REGISTER requests. A registrar is typically co-located with a proxy or redirect server and MAY offer location services.

5 3.0 Overview of SIP operation SIP is modeled after the simple mail transfer protocol (SMTP), the basis of , and the hypertext transfer protocol (HTTP), the basis of the Web. Like SMTP and HTTP, SIP is a textual client server protocol, in which the client issues requests and the server returns responses. In fact, SIP uses much of the syntax and semantics of HTTP, including its response code architecture, many message headers, and its overall operation. But unlike them, SIP can run on top of either TCP or UDP. Hence SIP supports multicasting, which enables group invitations and basic automatic call distribution (ACD) functions. SIP messages are of two types: Requests and Responses. The user agent client, on behalf of the user, issues requests and the responses will be received and processed by the user agent server. A SIP request consists of a request line, header fields and a message body. The various header fields contain information on call services, addresses, and protocol features. The body, opaque to SIP, can be defined in any format like SDP. SIP defines several methods, including INVITE, BYE, OPTIONS, ACK, REGISTER and CANCEL. INVITE method is used a user to a call. The header fields of this request contain the addresses of the caller and callee, subject of the call, call priority, call routing requests, caller preferences and desired features of the response. It also contains information on codecs, ports and protocols to be used for sending media to the caller. REGISTER method is used to convey the location information to a SIP server. It allows a user to tell a SIP server how to map an incoming address into an outgoing address that will reach that user. BYE method is used to terminate a connection between two users in a conference. OPTIONS method is used to solicit information about the capabilities of the callee. ACK method confirms the reliable message exchanges. CANCEL method terminates a pending request. It does not undo a completed call.

6 SIP responses are of two types : Provisional and Final. Provisional responses are issued by the server to indicate progress, but do not terminate a SIP transaction. Final responses terminate the transaction. The responses are as follows: 1XX Informational 2XX Success 3XX Redirection 4XX Client error 5XX Server error 6XX Global failure.

7 4.0 Typical Message Sequence Scenarios Ã&$6(Ã'LUHFWÃ&RPPXQLFDWLRQÃEHWZHHQÃ(QGÃ3RLQWVÃ port 3567 port INVITE 100 Trying 180 Ringing ACK RTP Flow BYE UAC UAS Madhu who is registered as wants to talk to she who is registered as Madhu knows the IP address of the callee and hence he sends an INVITE request to the destination. Madhu received trying and ringing response, before he receives OK response from the callee. Madhu then sends an ACK, confirming the call. RTP media flows between the callee and caller. Either madhu or she can terminate the call by sending a BYE request. The other party acknowledges with a response.

8 Ã&$6(Ã6,3Ã2SHUDWLRQÃLQÃWKHÃSUHVHQFHÃRIÃDÃ5HGLUHFWÃ6HUYHUÃ Ã UAC Redirect Server UAS INVITE 302 Moved temporarily ACK INVITE 100 Trying 180 Ringing ACK RTP Flow BYE She wants to talk to madhu. She sent an INVITE to the redirect server. Redirect server contacts the location server and infers that madhu is presently moved to another address It returns the response. UAC of She contacts the address returned by the redirect server. The rest will proceed as described in the first case.

9 Ã&$6( 6,3ÃRSHUDWLRQÃLQÃWKHÃSUHVHQFHÃRIÃDÃ3UR[\ÃVHUYHU Caller UA Proxy server Callee UA INVITE she INVITE 100 Trying 180 Ringing ACK ACK RTP Flow BYE Caller sends an INVITE request to the locally configured proxy server. The proxy finds the destination address, by contacting the location server. If the location server returns multiple addresses, the proxy sends multiple INVITEs to all those addresses simultaneously (Forking Proxy). The proxy returns Trying and Ringing responses to the caller UAC, while it waits for a response from the caller. If the caller sends response, the proxy returns the response to the callee. The callee then sends ACK to the proxy, which it will send to the caller end. The ACK can also be directly sent to the caller, bypassing the proxy. RTP media flows between the ends (Proxy won t come into picture here). The rest of the process is same as described in the previous case.

10 4.4 Example of a typical INVITE Request INVITE SIP/2.0 From: madhu To: test Call-ID: CSeq: INVITE Content-Type: application/sdp v=0 o=madhu IN IP s=voip i= a seminar on voice over internet protocol t= m= audio 5004 RTP/AVP SIP Header SDP Message The method starts with the word INVITE. The caller id and the current version of the SIP are given in the first line. The field From: contains the address of the caller, and the To contains that of the callee. Call-ID is the globally unique identifier (space and time) for the call. CSeq also contains a random identifier, which is unique for a particular SIP session (except for re-invitations, which contain a higher CSeq number). The field Content-Type gives the type of media description. The fields from, to, Call-ID together are called Call Leg. The Call Leg is unique throughout the SIP session. The SIP payload is described in SDP in this example. It contains information about the SDP version, owner details, subject name and details, start and stop times, and details about the media. The start and stop times are irrelevant in the case of two-way calls, but they carry meaning in the session announcements. The media details contain the type of media (audio/video), the port on which media will be transported and the various codecs codes that the user can support. The codes are given in the AVP (Audio Video Profile). IETF has recommended that the codes 0 to 5 (PCM, 1016, G.721, GSM, Unassigned, DVI4) are mandatory and all SIP end points must support them for the sake of interoperability. The SIP transactions are carried on a default port (5060) or any other random port as configured by the user. Similarly the media will be transported on a default port (5004) or any other unique port as configured by the user.

11 5.0 Advanced Services in SIP In addition to the above features, SIP can also support the following advanced features. Call forwarding unconditional, busy. Call transfer Caller ID Call holding 3-way conferencing and multi-party conferencing. Call return Call parking (with NOTIFY). Follow me Call Waiting IVR systems Multiple line presences. Camp on Call queuing Automatic call distribution Do no disturb Repetitive dialing. Station speed dialing. Last number redial Distinctive ringing.

12 6.0 Comparison between SIP and H.323 H.323 is another well known system, proposed by the ITU-T, for the multimedia communication systems. H.323 is a robust and umbrella standard, which includes many protocols like H.225 for call signaling, H.245 for call control, H.235 to incorporate security features and so on. For media transmission, both H.323 and IETF has adopted RTP/RTCP. Similarly for resource reservation, RSVP has been supported by both the systems. The pro s and con s of the two systems has been discussed below: SNo H.323 SIP 1. Robust but consumes more call set up time (7.5 times RTT for version 1) Simple, Scalable and extensible (Requires only 1.5 times RTT) 2 Requires about twelve packets for call set-up Requires about four packets for call set-up. 3 Provides floor control within a session Cannot provide 4 Has more elaborate capability exchange (H.245) Minimal capability exchange (SDP & AVP): Enough for IP Telephony. 5 Provides a multipoint controller for conferences Not required for SIP multicast conferences. 6 Requires both TCP and UDP during the call-set Basically runs on UDP. up Reliability achieved through retransmissions. Supports TCP also, if UDP is not supported. 7 Implementation is complex and time taking Easy to implement 7.0 Conclusion Session Initiation Protocol (SIP) is used for initiation, control, and termination of multimedia conferences. SIP follows some of the latest and most widely used protocols like SMTP and HTTP. The strength of SIP lies in its simplicity, extensibility, scalability, support rich mobility, address resolution, and naming services. SIP can also be used for call screening, personal mobility, and residential line services. SIP system can be made interoperable with other IP telephony systems like H.323.

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