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Session Initiation Protocol

Session Initiation Protocol. Team Members: Manjiri Ayyar Pallavi Murudkar Sriusha Kottalanka Vamsi Ambati Girish Satya LeeAnn Tam. Agenda. Introduction to SIP Overview of functionality SIP components SIP protocol layers SIP call flows SIP concerns Demo Conclusion. Introduction.

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Session Initiation Protocol

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  1. Session Initiation Protocol Team Members: Manjiri Ayyar Pallavi Murudkar Sriusha Kottalanka Vamsi Ambati Girish Satya LeeAnn Tam

  2. Agenda • Introduction to SIP • Overview of functionality • SIP components • SIP protocol layers • SIP call flows • SIP concerns • Demo • Conclusion

  3. Introduction Session Initiation Protocol (SIP) • application layer signaling protocol • used to create, manage and terminate sessions in an IP based network. RFC : 3261

  4. Circuit switched Network • Circuit is fully established between the two devices before data is sent. • Less efficient since much of the bandwidth is wasted.

  5. Packet switched network • No fixed path is established between devices • Data broken into packets. • Packets may take multiple paths to reach the destination device. • More efficient.

  6. SIP applications • VoIP • Video Conferencing • Instant Messaging

  7. Multimedia session in a packet switched network A typical real-time multimedia session requires • Session management : Users may move from terminal to terminal with different capabilities. To set up communication session between two or more users, a signaling protocol is needed. • Media transport : RTP is used for transmitting real-time data like audio and video. • End-to-End delivery : Underlying IP layer which connects the whole world.

  8. SIP functionality SIP is limited to only the setup, modification and termination of sessions. • Establishment of user location • Feature negotiation • Call management • Changing features while a session is in progress All of the other key functions are done with other protocols

  9. SIP components The key components in a SIP network are • SIP Clients : SIP Phones (User-Agents) • SIP servers • SIP PSTN gateways • Application servers (such as media servers)

  10. SIP Network

  11. SDP codecs SIP RTSP RTP DNS(SRV) TCP UDP IP Ethernet Where’s SIP • Application • Transport • Network • Physical/Data Link

  12. SIP – Protocol Layers TransactionUser Transaction Transport Syntax and Encoding

  13. SIP - Messages start-line *message-header CRLF [ message-body ] start-line = Request-Line / Status-Line

  14. INVITE  Requests a session • ACK   Final response to the INVITE • OPTIONS  Ask for server capabilities • CANCEL  Cancels a pending request • BYE   Terminates a session • REGISTER  Sends user’s address to server

  15. SIP - Responses • 1XX  Provisional  100 Trying • 2XX  Successful  200 OK  • 3XX  Redirection  302 Moved Temporarily  • 4XX  Client Error  404 Not Found • 5XX  Server Error  504 Server Time-out • 6XX  Global Failure  603 Decline

  16. SIP Call Scenarios Session Registration Establishment , Termination RFC 3665

  17. User A Registrar Server Location Server Register sip:bob@lab.acme.com bob@lab.acme.com Contact 10.18.2.4 200 - OK Registration binds a particular device Contact URI with a SIP user Address of Record.

  18. SIP call with Proxy Server Host2.com proxy Host1.com proxy Alice Bob Invite F1 Invite F2 Invite F4 100 Trying F3 100 Trying F5 180 Ringing F6 180 Trying F7 180 Trying F8 200 OK F9 200 OK F10 200 OK F11 ACK F12 Media Session Bye F13 200 OK F14

  19. SIP INVITE INVITE sip:bob@biloxi.com SIP/2.0 Via: SIP/2.0/UDP pc33.atlanta.com;branch=z9hG4bK776asdhds Max-Forwards: 70 To: Bob <sip:bob@biloxi.com> From: Alice <sip:alice@atlanta.com>;tag=1928301774 Call-ID: a84b4c76e66710@pc33.atlanta.com CSeq: 314159 INVITE Contact: sip:alice@pc33.atlanta.com Content-Type: application/sdp Content-Length: 142

  20. SIP - Response SIP/2.0 200 OK Via: SIP/2.0/UDP server10.biloxi.com ;branch=z9hG4bKnashds8;received=192.0.2.3 Via: SIP/2.0/UDP bigbox3.site3.atlanta.com ;branch=z9hG4bK77ef4c2312983.1;received=192.0.2.2 Via: SIP/2.0/UDP pc33.atlanta.com ;branch=z9hG4bK776asdhds ;received=192.0.2.1 To: Bob <sip:bob@biloxi.com>;tag=a6c85cf From: Alice <sip:alice@atlanta.com>;tag=1928301774 Call-ID: a84b4c76e66710@pc33.atlanta.com CSeq: 314159 INVITE Contact: <sip:bob@192.0.2.4> Content-Type: application/sdp Content-Length: 131

  21. SIP Concerns • Security • Authentication of signaling data using HTTP digest authentication • TLS usage (over TCP) • Usage of IPSec (SIP VPN Scenario) • Use SecureRTP for Media • Use S/MIME to enable mechanisms like public key distribution, authentication, integrity and confidentiality of SIP signaling data

  22. SIP Concerns…contd • Quality of Service • Latency, network delays (upper bound is 150ms) • Jitter ( refers to non-uniform delays ) • Packet Loss • Power Failure and Backup Systems • Interoperability

  23. Demo • User Agents used : Yahoo Messenger • Call Scenarios Covered: • Register • Call Establishment • Call Termination

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