1. 1 Kommunikatsiooniteenuste arendus IRT0080 Loeng 8 Avo Ots telekommunikatsiooni õppetool, TTÜ raadio- ja sidetehnika inst. avo.ots@ttu.ee

2. 2 A signaling protocol for setting up multimedia sessions between endpoints Fundamental shift from PSTN : infrastructure consists of software on standard servers –SIP designed in line with other Internet protocols by the IETF –Uses overlay control network consisting of SIP Proxies to route SIP messages : Media path (RTP/UDP) decoupled from signaling –name@domain addressing; message syntax similar to HTTP SIP (Session Initiation Protocol) SIP proxy SIP signaling RTP/UDP packets

3. 3 Protocols Session Initiation Protocol – for signaling Reservation Protocol – for reserving network resources Real-Time protocol – Transporting real-time data and providing QoS feedback Real-Time Streaming protocol – for controlling delivery of streaming media Session Advertising Protocol – advertising multimedia sessions via multicast Session Description Protocol – to describe multimedia sessions

4. 4 Session Initiation Protocol Application layer protocol for creating, modifying and terminating multimedia sessions Supports unicast, mesh and multicast conferences, and combination of these modes Similar to HTTP, Client-Server protocol, and text- based Provides reliability without assuming anything about the lower layer transport protocols

5. 5 Components User Agents –User Agent Client –User Agent Server Network Servers –Registration Servers –Proxy Servers Stateful Proxy Server Stateless Proxy Servers –Redirect Servers

6. 6 Parser Sip messages are sent as text (Unicode characters) The Sip message grammar is given in the augmented BNF-form (RFC 2543) Syntax tree builder (JTBGJ) Parser Generator (Javacc)

7. 7 Protocol Building blocks User Agent A Registration Server User Agent B Proxy Server Location Server B’s current location Registration Server

8. 8 Registration SIP Client sends REGISTER message to the Registration server. Indicates the physical location REGISTER messages can be multicast (“sip.mcast.net” – 224.0.1.75) Serves as a simple locator service. REGISTER sip:belltell.com SIP/2.0 Via: SIP/2.0/UDP saturn.belltell.com From: sip:watson@belltell.com To: sip:watson@belltel.com CSeq: 1 REGISTER Call-ID: 70710@saturn.belltell.com Contact: sip:watson@globe..belltell.com:3890 Expires: 7200

9. 9 Name Translation, UserLocation SIP Addressing by a Sip URL (user@host:port)user@host:port Resolves host name through –DNS server –Checks if domain name refers to a host –Mail exchange host User at a specific host derived through zero or more translations If called party is not at SIP server, can be directed or proxied (personal mobility)

10. 10 Session invitation Caller sends the request directly to UAS or to the nearest proxy server Proxy server uses the locator service to determine the next hop server Proxy forwards the request towards the callee Redirect server simply returns the alternate addresses INVITE sip:watson@ieee.org SIP/2.0 Via: SIP/2.0/UDP c.belltell.com From: sip:a.g.bell@belltell.com To:sip:watson@ieee.org Call-ID:23534@c.belltell.com23534@c.belltell.com Cseq:1 INVITE

11. 11 Session Invitation (cont.) If there are more than one locations to be contacted, the proxy forks the requests to all the locations. The callee’s response is propagated backwards the servers ( uses the Via header ) Stateful Proxy maintains the state according to the Call-ID. So forwards responses accordingly The caller sends ACK request following the response.

12. 12 Feature negotiation Allows a group of end systems to agree on the media to exchange and their respective parameters such as encodings. INVITE request has a listing of media types and associated encodings that the calling party is willing Called party responds with a subset of media types and encodings that it is willing to use Feature changes –Subsequent INVITE request –OPTIONS request

13. 13 200 (OK) INVITE 180 (Ringing) 200 (OK) ACK RTP MEDIA PATH BYE 200 (OK) Call Teardown Media Path Call Setup INVITE Proxy 200 (OK) INVITE 180 (Ringing) ACK BYE 200 (OK) Proxy User Agent VoIP Call setup : SIP message flow

14. 14 Motivation Collaborative Apps SameTime Notes Calendar/email (SIP-enabled) Services/ Infrastructure Corporate VoIP/data network network (VoIP trial) VoIP infrastructure Enhancements (conferencing, context) Web Browser VoIP enabled portals Client-side SIP service In-house conferencing service (IP/ PSTN phones) Distributed application scenarios, e.g. networked games Game servers, IM servers

15. 15 Application Interface SIP Service Web Browser IM Agent Email Client Other Apps (Multi-player Games, …) Wrapper 1 Wrapper 2 Wrapper 3 SIPStackwithpub/sub Client Laptop IP Phone SoftPhone PSTN/ cell phone

16. 16 LINGID http://www.wifi.ee/foorum2/viewtopic.php?t= 1083 http://www.packetizer.com/voip/sip/ http://www.tech-invite.com/

17. 17 Lingid (1) http://www.eventhelix.com/RealtimeMantra/Congesti onControl/queueing_theory.htm http://www.tmcnet.com/voip/conference/m05/sip- workshop.aspx http://en.wikipedia.org/wiki/SIP_trunking http://en.wikipedia.org/wiki/Session_Initiation_Protoc ol http://en.wikipedia.org/wiki/IP_Multimedia_Subsyste m

18. 18 Lingid (2) http://www.ietf.org/rfc/rfc4740.txt?number=4740 http://www.cisco.com/en/US/products/ps6638/product s_data_sheet09186a00804fe332.html http://www1.cs.columbia.edu/~kns10/talks/belllab s.ppt http://www1.cs.columbia.edu/~kns10/talks/