SIP Overview(1) - History SIP ( Session initiation protocol) Work began in 1995 Developed in SIP Working Group in IETF Proposed standard RFC2543, February 1999 New Version of SIP - RFC 3261, June 2002 Split from MMUSIC (Multiparty Multimedia Session Control), Sep. 1999
SIP Overview(2) - Main Functions SIP is an application-layer control (signaling) protocol for : initiating sessions Find the user ’ s current location Carry session descriptions Modifying sessions Terminating sessions
SIP Overview(3) - What is a session? Refers to an active connection between two computers for the purpose of communicating and transferring information Internet telephone calls multimedia conferences Instant Messaging However it ’ s not limited to the above
SIP Overview(4) SIP is similar to HTTP : both use a request/response model both are text-based much of SIP ’ s message and header field syntax is identical to HTTP/1.1. Works independent of the underlying network transmission protocol and indifferent to media
SIP Overview(5) Scalability Functionality such as proxying, redirection, location, or registration can reside in different physical servers. Distributed functionality allows new processes to be added without affecting other components.
SIP Overview(6) Interoperability An open standard Can implement to communicate with other SIP based products
SIP Overview(7) Mobility The user can be using a PC at work, PC at home,wireless phone, IP phone, or regular phone. Users must register/Update their current location. Supports user mobility by proxying or redirecting requests to a user ’ s current location. Proxy servers will forward calls to the user ’ s current location.
Integration with IETF Protocols SIP forms only part of an overall IP telephony system Other IETF protocol standards are used to build a fully functioning VoIP system. example: RSVP - to reserve network resources. RTP (Real Time Transport Protocol) - to transport real time data RTSP (Real Time Streaming Protocol) - for controlling delivery of streaming media. RTCP (Real Time Transport Control Protocol) – provides information about the quality of the communication.
SIP Capabilities(1/2) Determine location of target points – Support address resolution, name mapping, call redirection Determine media capabilities – SIP uses Session Description Protocol (SDP) for this Determine availability – returns a message why the remote party cannot be contacted Establish a session between end points – also support mid call changes, changes of media characteristics or codec
SIP Capabilities(2/2) Permits interaction between devices via signaling messages These messages can: Register a user with a system Invite a user to join an interactive session Establish a media stream between 2 or more end points Terminate a session
SIP Component – User Agent User Agent Client (UAC)- Initiate SIP Request User Agent Server (UAS)-Accepts or rejects call Phones – acts as UAC or UAS Implemented in Hardware or Software Components Includes softphones, sip ip phones
SIP Component – Proxy Server Heart of SIP network that contains all service logic. Receives SIP messages, forwards to next SIP UAS. Interprets, rewrites or translates a request message before forwarding it.
SIP Component – Registrar Accept registration requests from users Maintains user ’ s whereabouts at a Location Server Typically co-located with a proxy server or a redirect server and may offer location services
SIP Addressing The SIP address is identified by a SIP URL,in the format: user@host. Email-like identifier of the form Examples of SIP URLs: sip:email@example.com sip:firstname.lastname@example.org sip:email@example.com
SIP Messages The message syntax is identical to HTTP/1.1(RFC2616). Two kinds of SIP messages SIP-message = Request | Response
SIP Messages - Requests INVITE – Initiates a call by inviting user to participate in session. ACK - Confirms that the client has received a final response to an INVITE request. BYE - Indicates termination of the call. CANCEL - Cancels a pending request. REGISTER – Registers the user agent. OPTIONS – Used to query the capabilities of a server.
SIP Messages - Responses 1xx - Informational Responses 180 ringing 2xx - Successful Responses 200 OK 3xx - Redirection Responses 302 Moved Temporarily 4xx - Request Failure Responses. 404 Not Found 5xx - Server Failure Responses. 503 Service Unavailable 6xx - Global Failures Responses. 600 Busy Everywhere All responses, except for 1XX, are considered final responses
Message Headers Provide further information about the message E.g., To : header in an INVITE The called party From : header The calling party Four main categories General, Request, Response, and Entity headers
Header Fields General Header Be applied to both request and response messages Entity Header Define information about the message body Request Header Allow the client to pass additional information about the request Response Header Allow the server to pass additional information about the response
SIP Headers – From A field required in all requests and response messages Provides identity of request ’ s initiator Example: From: Laura Brown
SIP Headers – To Provides identity of the intended recipient of the request Example: To: Bob Johnson ;
SIP Headers – Call-ID(1/2) Provides a globally unique identifier to distinguish specific invitations Typically uses a 32-bit cryptographically random numbers Example: Call-ID: firstname.lastname@example.org
SIP Headers – Contact(1/2) A Contact header provides a URL where the user can be reached directly. This feature is important because it offloads SIP servers that do not need to be in the signalling path after routing the first INVITE. Example: Contact: Bob Johnson
SIP Header - Via History of message ’ s path through network(s) Helps to prevent looping and ensures replies route back to originator Indicates the used transport protocol, ip address and port of sender Via: SIP/2.0/UDP 192.168.6.21:5060
SIP Header – Content-Type Provides information about media type of message body Content-Type: application/sdp
SIP Header – CSeq(1/2) CSeq or command sequence: Needed in both request messages as well as response messages Need to increment this when a user with the same Call-ID wants to send different SIP methods or content ( Except ACK and Cancel) When sending responses to requests, CSeq should be the same The numerical part of the Cseq is used to order different requests within the same session. CSeq: 1 INVITE
Reliable Transmission of Responses(1/2) Final responses are transmitted reliably between server and client, using retransmissions or a reliable transport protocol to ensure delivery. Provisional responses may either be received by the client or be lost in the network.
SIP is gaining acceptance in the industry Open Source projects are taking the lead in SIP implementations New generation of services are already being offered Staying with single a vender solution is currently workable Interoperability between vender products is still in early stages
Reference RFC 2543(SIP/1.0) RFC 3261(SIP/2.0) RFC 2327(SDP) Related Material provided by