html5-img
1 / 48

Kommunikatsiooniteenuste arendus IRT0080

Kommunikatsiooniteenuste arendus IRT0080. Loeng 4 Avo Ots telekommunikatsiooni õppetool, TTÜ raadio- ja sidetehnika inst. avo.ots@ttu.ee. SIP Overview(1) - History. SIP ( Session initiation protocol) Work began in 1995 Developed in SIP Working Group in IETF

erling
Télécharger la présentation

Kommunikatsiooniteenuste arendus IRT0080

An Image/Link below is provided (as is) to download presentation Download Policy: Content on the Website is provided to you AS IS for your information and personal use and may not be sold / licensed / shared on other websites without getting consent from its author. Content is provided to you AS IS for your information and personal use only. Download presentation by click this link. While downloading, if for some reason you are not able to download a presentation, the publisher may have deleted the file from their server. During download, if you can't get a presentation, the file might be deleted by the publisher.

E N D

Presentation Transcript

  1. Kommunikatsiooniteenuste arendusIRT0080 Loeng 4Avo Otstelekommunikatsiooni õppetool, TTÜ raadio- ja sidetehnika inst.avo.ots@ttu.ee

  2. 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

  3. 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

  4. 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

  5. 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

  6. 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.

  7. SIP Overview(6) • Interoperability • An open standard • Can implement to communicate with other SIP based products

  8. 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.

  9. 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.

  10. Related Protocols

  11. 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

  12. 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

  13. SIP Network Architecture

  14. 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

  15. User Agents

  16. 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.

  17. Scenario with Proxy Server

  18. SIP Component – Redirect Server • Return routing information to the originating endpoint. • Does not accept or terminate calls • Does not initiate its own SIP request

  19. Scenario with Redirect Server

  20. 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

  21. Scenario with Registrar

  22. Invitation for SIP Proxy Server

  23. Invitation for Redirect Server

  24. 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:hostname@ntut.edu.tw • sip:hostname@192.168.10.1 • sip:14083831088@vovida.org

  25. SIP Messages • The message syntax is identical to HTTP/1.1(RFC2616). • Two kinds of SIP messages SIP-message = Request | Response

  26. 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.

  27. 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

  28. 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

  29. 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

  30. SIP Headers

  31. SIP Headers – From • A field required in all requests and response messages • Provides identity of request’s initiator • Example: • From: Laura Brown <sip:Laura.Brown@university.com>

  32. SIP Headers – To • Provides identity of the intended recipient of the request • Example: • To: Bob Johnson <sip:Bob.Johnson@company.com>;

  33. 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: ges456fcdw21lkfgte12ax@workstation1234.university.com

  34. SIP Headers – Call-ID(2/2)

  35. 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:Bob@140.124.181.227>

  36. SIP Headers – Contact(2/2)

  37. 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

  38. SIP Header – Content-Type • Provides information about media type of message body • Content-Type: application/sdp

  39. 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

  40. SIP Header – CSeq(2/2)

  41. An example of SIP Request

  42. An example of SIP Response

  43. 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.

  44. Reliable Transmission of Responses(2/2)

  45. Reliable Transmission of Requests(1/2)

  46. Reliable Transmission of Requests(2/2)

  47. Summary • 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

  48. Reference • RFC 2543(SIP/1.0) • RFC 3261(SIP/2.0) • RFC 2327(SDP)

More Related