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Presented By Team Netgeeks SIP Session Initiation Protocol
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AGENDA Introduction Applications Overview of operation ○ Protocol Elements ○ Requests ○ Responses SIP based services Security
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Introduction SIP Session Initiation Protocol, RFC 3261 A signaling, presence and instant messaging protocol. Developed to set up, modify, and tear down multimedia sessions, request and deliver presence and instant messages over the Internet. SIP allows two end points to establish media sessions with each other.
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Introduction The main signaling functions of the protocol are as follows: Location of an end point. Contacting an end point to determine willingness to establish a session. Exchange of media information to allow session to be established. Modification of existing media sessions. Tear-down of existing media sessions.
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Introduction Also been extended to request and deliver presence information (on-line/off-line status and location information such as that contained in a “buddy” list, as well as instant message sessions. These functions include: Publishing and uploading of presence information. Requesting delivery of presence information. Presence and other event notification. Transporting of instant messages.
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SIP applications Setting up voice-over-IP calls Setting up multimedia conferences Event notification (subscribe/notify) a IM and presence Text and general messaging Signaling transport
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SIP SIP is based on an HTTP-like request/response transaction model. SIP works with both IPv4 and IPv6. SIP uses the well-known port number of 5060 or 5061. It can use either TCP or UDP as transport protocol.
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Overview of operation The basic functions of SIP: location of an end point, signal of a desire to communicate, negotiation of session parameters to establish the session, and teardown of the session once established. Incorporates elements of two widely used Internet protocols: Hyper Text Transport Protocol (HTTP) and Simple Mail Transport Protocol (SMTP) used for e-mail. From HTTP, SIP borrowed a client-server design and the use of URLs and URIs. From SMTP, SIP borrowed a text-encoding scheme and header style. SIP identity Uniform Resource Identifier (URI) called a SIP URI sip:bob@macrosoft.com Secure URI, called a SIPS URI sips:bob@macrosoft.com
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Basic topology Alice Wonder- Land Macro- Soft Bob INVITE OK RTP INVITE OK Note that the media is always end-to-end and not through the proxy.
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SIP Protocol Elements SIP has the following entities, each with different function. 1. SIP Terminal - Supports the real time, two-way communication with other SIP entities. 2. SIP User Agent - user agents are the endpoints of the call. User Agent Client (UAC) initiating the call User Agent Server (UAS) answering the call. Receives and responds to SIP requests and can accept, refuse or redirect the call. The User Agent software switches between the UAC and UAS modes on a message-by-message basis depending on what is going on. The User Agents can be handsets or desktop applications.
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SIP Protocol Elements (cont’d) 3. SIP Network Server - Handles the signaling associated with multiple calls and allows peer-to-peer calls to be made using client-server protocol. Main function is to provide name resolution and user location, and to pass on messages to other servers using next- hop routing protocols. There is more than one type of server: the Proxy Server, Redirect Server, and the Registrar Server. Proxy Servers - Network hosts acting as both clients and servers to other entities. The job is to ensure requests are routed to appropriate entity identified by a SIP Uniform Resource Identifier (URI). The Proxy servers can operate in two different modes: The SIP stateful Proxy server and the SIP stateless Proxy server. Redirect Servers receive SIP requests and send response to zero or more addresses. The first location to answer takes the call. Redirect servers do not initiate SIP requests or accept SIP calls. Registrar Servers accept registration requests. These servers maintain the databases that contain location information of all user agents registered with a particular SIP domain, thereby enabling the users to update their location and policy information.
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SIP - Requests INVITE - initiate call REGESTER - register with location service Other Methods: ACK - confirm final response BYE - terminate (and transfer) call
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Session Initiation and Tear Down INVITE F1 INVITE F2 INVITE F4 TRYING F5 TRYING F3 RINGING F7 RINGING F8 200 OK F9 200 OK F10 200 OK F11 ACK F12 BYE F13 200 OK F14 Media Stream RINGING F6 Alice’s Softphone Alice’s Proxy Server Bob’s Proxy Server Bob’s Softphone
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Sample Request Format INVITE sip:bob@macrosoft.com SIP/2.0bob@macrosoft.com Via: SIP/2.0/UDP pc33.wonderland.com; Max-Forwards: 70 To: Bob bob@macrosoft.com From:Alice ;tag=1928301774sip:alice@wonderland.com>;tag=1928 Call-ID: a84b4c76e66710@h3.wonderland.coma84b4c76e66710@h3.wonderland.com CSeq: 314159 INVITE Contact: alice@h3.wonderland.com Content-Type: application/sdp Content-Length: 142
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Sample Response Format SIP/2.0 200 OK Via: SIP/2.0/UDP server10.macrosoft.com ;branch=z9hG4bKnashds8;received=192.0.2.3 Via: SIP/2.0/UDP bigbox3.site3.wonderland.com;received=192.0.2.2 Via: SIP/2.0/UDP h3.wonderland.com;branch=z9hG4bK776asdhds ;received=192.0.2.1 To: Bob ;tag=a6c85cfbob@m.macrosoft.com From: Alice ;tag=1928301774alice@wonderland.com Call-ID: a84b4c76e66710@h3.wonderland.coma84b4c76e66710@h3.wonderland.com CSeq: 314159 INVITE Contact: Content-Type: application/sdp Content-Length: 131
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SIP Responses
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SIP Response Routing
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SIP Transportation SIP operates over any packet network, reliable or unreliable choices: UDP: most common – low state overhead – small max. packet size TCP: can combine multiple signaling flows over one link – use with SSL – connection setup overhead – HOL blocking for trunks SCTP: new protocol – no HOL blocking – fallback address (but SRV provides this already) – connection setup overhead
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SIP – based services Call forwarding: basic INVITE behavior (proxy/redirect) Call transfer: REFER method Call hold: set media address to 0.0.0.0 – can be done individually per media Caller id: From, plus extensions
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SIP – based services ( Call Tx )
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SIP presence protocol
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Security Threats spoofing From in REGISTER: call redirection spoofing From in INVITE: bypass call filtering snooping media packets billing confusion (identifier munging) denial-of-service attack
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Security (cont’d) Layer/mechanismApproachcharacteristics Network LayerIPSechard to configure. Transport LayerTLS SIP INVITEbasic/digestShared secrets with random parties. SIP REGISTERbasic/digestsecuring headers? Basic (plaintext password) and digest (challenge- response) are very similar to HTTP security mechanisms.
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References http://www.faqs.org/rfcs/rfc3261.html http://www.faqs.org/rfcs/rfc3261.html SIP – Understanding The Session Initiation Protocol by Alan B.Johnston http://www.cs.columbia.edu/~hgs/teaching/ai s/slides/2003/sip_long.pdf http://www.cs.columbia.edu/~hgs/teaching/ai s/slides/2003/sip_long.pdf
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