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Understanding SIP (I)

Understanding SIP (I). Outlines. Something About IETF It’s IP Telephony IP Telephony Basics Protocol Zoo SIP Signaling Reference. Something about IETF. Internet Engineering Task Force ( www.ietf.org ) IETF’s bussiness Standardization Procedure ( RFC 2026 )

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Understanding SIP (I)

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  1. Understanding SIP (I)

  2. Outlines • Something About IETF • It’s IP Telephony • IP Telephony Basics • Protocol Zoo • SIP Signaling • Reference

  3. Something about IETF • Internet Engineering Task Force (www.ietf.org) • IETF’s bussiness • Standardization Procedure ( RFC 2026 ) • Advantages of the IETF Standardization Process • Related IETF Working Groups • SIP • IPTEL

  4. Appeals of IP Telephony • Saving • Lower QoS • Internet Service Integration • Distribution Games • Instant Messaging • In IP, you are your own master. • Open service market • Programmability

  5. What Protocols Are Needed? • Signaling protocol • SIP / SDP • Media protocol • RTP • Transport protocol • TCP / UDP • Supporting Protocol • DNS • Diameter

  6. Protocol ZOO http://www.cs.columbia.edu/~hgs/internet/

  7. SIP Signaling

  8. Session Initiation Protocol • SIP is end-to-end, client-server session signaling protocol • Arbitrary services built on top of SIP • Features: • Text-based Encoding • Programmability • History

  9. SIP-General Purpose Presence Protocol • SIP is not limited to Internet telephony • Suitable for applications having a notion of session • Network games • Video conferencing, etc.

  10. SIP Workhorses • SIP Proxy Server • Relays call signaling • Operates in a transactional manner • SIP Redirect Server • Redirects callers to other servers • SIP Registrar • Accept registration requests from users • Maintain user’s whereabouts at a Location Server

  11. SIP Addresses • SIP gives you a globally reachable address • URLs used as address data format; examples: • sip:jiri@iptel.org • sip:voicemail@iptel.org?subject=callme • Sip:sales@hotel.xy; geo.position=48.54_-123.84_120 • Must include host, may include user name, port number, parameters, etc. • Address space unlimited • Non-SIP URLs can be used as well (mailto:...)

  12. REGISTER sip:iptel.org SIP/2.0 From:sip:jiri@iptel.org To:sip:jiri@iptel.org Contact:<sip:195.37.78.173> Expires:3600 jiri@195.37.78.173 SIP/2.0 200 OK SIP Registration Location Server SIP Registrar (domain: iptel.org)

  13. jiri ? INVITE sip:jiri@195.37.78.173 From:sip:Caller@sip.com To:sip:jiri@iptel.org Call-ID:345678@sip.com INVITE sip:jiri@iptel.org From:sip:Caller@sip.com To:sip:jiri@iptel.org Call-ID:345678@sip.com jiri@195.37.78.173 SIP/2.0 200 OK SIP/2.0 200 OK ACK sip :jiri@195.37.78.173 SIP Operation in Proxy Mode Location Server SIP Proxy Server Caller@sip.com jiri@195.37.78.173

  14. Proxy Server Functionality • Serve as rendezvous point at which callees are globally reachable. • Perform routing function • Allow the routing function to be programmable. Arbitrary logic may be built on top of the protocol. • Forking

  15. Proxy Chaining • There may be also cases when a local outbound proxy may be involved • In general, Server may be arbitrarily chained. • Servers have to avoid loops and recognize spirals.

  16. Proxy Chaining - Example

  17. “Stateful” Proxy Refers to Transactions SIP state forgotten as soon as transactio over Unless route recording is used, Bye may take a completely different Path to destination in Contact: header field BYE

  18. Callee ? Callee@home.com INVITE sip:Callee@example.com 302 Moved Temporarily Contact: Callee@home.com ACK sip:Callee@example.com INVITE sip:Callee@home.com SIP/2.0 200 OK ACK sip:Callee@home.com SIP Operation in Redirect Mode Location Server Caller@sip.com Callee@home.com SIP Redirect Server

  19. SIP Server - Proxy v.s Redirection • A SIP Server may either proxy or redirect a request. • Redirection useful if a user moves or changes her provider. • Proxy useful if forking AAA, firewall control needed.

  20. SIP RFC2543 Methods

  21. Response Status Line • SIP-Version SP Status-Code SP Reason-Phrase CRLF • Status-Code = SIP/2.0 SP 180 SP Ringing CRLF

  22. SIP Message Structure

  23. Reference • Understanding SIP, GMD Fokus • Introduction to SIP in Chunghwa Telecom Co. Ltd, Wen-Ping Lai, III • RFC 3261 (SIP: Session Initiation Protocol )

  24. Understanding SIP (II)

  25. Outlines • Session Description Protocol (SDP) • SIP Protocol Design • Real Time Transport Protocol (RTP) • Real-Time Transport Control Protocol (RTCP) • Media Path != Signaling Path • Programming SIP • SIP && QoS Control • To Be Continuing…….

  26. Session Description Protocol (SDP) • A well defined format for conveying sufficient information to discover and participate in a multimedia session – RFC 2327 • Session Announcement Protocol (SAP) • Periodically multicasting an announcement packet to a well known multicast address and port – RFC 2974

  27. Session Description Protocol (SDP) • SDP is a data format rather than a protocol • SDP include: • Session name and purpose • Times the session is active • The media comprising the session • Information to receive those media (addresses, ports, formats and so on) • Information about the bandwidth to be used by the conference • Contact information for the person responsible for the session

  28. Session Description Protocol (SDP) • Session description • v= (protocol version) • o= (owner/creator and session identifier). • s= (session name) • i=* (session information) • u=* (URI of description) • e=* (email address) • p=* (phone number) • c=* (connection information - not required if included in all media) • b=* (bandwidth information) • z=* (time zone adjustments) • k=* (encryption key) • a=* (zero or more session attribute lines)

  29. Session Description Protocol (SDP) • Time description • t= (time the session is active) • r=* (zero or more repeat times) • Media description • m= (media name and transport address) • i=* (media title) • c=* (connection information - optional if included at session-level) • b=* (bandwidth information) • k=* (encryption key) • a=* (zero or more media attribute lines)

  30. Session Description Protocol (SDP)Example • v=0 • o=sisalem 28908044538 289080890 IN IP4 193.175.132.118 • s=SIP Tutorial • e=sisalem@fokus.gmd.de • c=IN IP4 126.16.69.4 • t=28908044900 28908045000 • m=audio 49170 RTP/AVP 0 98 • a=rtpmap:98 L16/11025/2

  31. SIP Protocol Design • Infrastructure follows IP state model • Most intelligence and state in the end-devices • Network core maintains at most transactional state • Network edge may maintain session state • Benefits: • memory and CPU consumption low in servers, reliability and scalability high (no single point of failure) • UDP Support • faster set-up, less state • Idempotent INVITEs • SIP requests are said to be idempotent , i.e., receiving more than one copy of a request does not change the server state

  32. SIP Protocol Design Extensibility • SIP designers took lesson from HTTP • Self-identifying Attribute-Value-Pairs (AVPs) followed by separators (EoL) • best-effort: receivers ignore unknown AVPs and skip to next separator • SDP support compulsory, arbitrary MIME payloads may be included (JPEG, ISUP, charging info, Multipart, ...) • transparent proxying

  33. SIP Protocol Design Extensibility (Cont.) • SIP designers took lesson from HTTP (cont.) • classes of status codes (1xx in-progress, 2xx success, 3xx forwarding, ...) • guidance on designing new extensions provided (draft-ietf-sipguidelines) • Capability inquiry with OPTION -- returns supported methods (Allow), media types (Accept), compression methods (Accepted-Encoding), Supported (supported features)

  34. SIP Protocol Design IP Based Multimedia Communication • SIP mainly establishes the IP addresses and port numbers at which the end systems can send and receive data • SIP does not transport data and does not depend on a certain compression • Data packets most probably do not follow the same path as the SIP packets

  35. SIP Protocol Design IP Based Multimedia Communication • Audio/Video samples are digitized, compressed and sent in UDP packets • Compression schemes use limitations of human ears/eyes to reduce bandwidth • Reduce audio bandwidth using silence suppression • Reduce video bandwidth using motion detection

  36. Compression Codecs more http://www.cs.columbia.edu/~hgs/(audio/video)

  37. Real Time Transport Protocol (RTP) • Standardized by the IETF and used by ITU-T as well • Designed to be scalable, flexible and separate data and control mechanisms Payload

  38. Real Time Transport Protocol (RTP) • Provides information for: • Media content type • Talk spurts • Sender identification • Synchronization • Loss detection • Segmentation and Reassembly • Security (encryption)

  39. RTP Header V:Version P:Padding for encryption X:Extension bit Payload type:Audio/Video encoding method Sequence number:Number of packet increased by one for each new packet Timestamp:Different fixed value for each compression type SSRC:A random number identifying the source (unique per source)

  40. Real-Time Transport Control Protocol (RTCP) • Exchange information about losses and delays between the end systems • Measure QoS • Packets sent in intervals determined based on number of end systems and available bandwidth

  41. Real-Time Transport Control Protocol (RTCP) • Sender Reports: Information about sent data, synchronization timestamp • Receiver Reports: Information about received data, losses, jitter and delay • Source Description: Name, Email, Phone, Identification • Bye: Explicit leave indication • Application defined parts: Parts for experimental functions

  42. Media Path != Signaling Path Location server Signing IP hop Proxy server Media

  43. Media Path != Signaling Path • SIP proxies can not usually control media path as there is split between signaling and media. • IP, DiffServ, and RSVP are the protocols for communication between end-devices and the network. • Attempts to manipulate media flows in the middle of path will difficult and may tend to fail: • A proxy does not know all IP hops along an end-to-end media path • Hops may belong to foreign administrative domains. • Signaling and media transport (possibly w/QoS) are two different businesses. • A SIP proxy may be located far apart from media path.

  44. Media Path != Signaling Path • For generality, extensibility and performance purposes, proxies do not parse SDP. • Even if they did, their operation might result in failure as new extensions (e.g., new codecs) • Even with SDP knowledge, proxies do not know entire media flow selectors -- SDP indicates only destination address of media streams. • SDP may be encrypted. • Unless route recording used, subsequent SIP requests (including ACK w/SDP) may take completely different path. • Exception to the rule: firewall control • Better than embedded ALGs • Firewalls located in the same administrative domain as a call party and its SIP proxy • The construct still suffers from shortcomings listed previously

  45. Programming SIP • Users and third parties may program • SIP follows HTTP programming model • Mechanisms suggested in IETF: CGI, Call Processing Language (CPL), Servlets • CPL – RFC 2824 • SIP CGI – RFC 3050

  46. Programming SIP (Cont.)Where may signaling services live ? • Some services have to live in the network: • Call distribution • Services for dial-up users without always-on IP connectivity • Some services can be implemented in both places: • Forward on busy • Some services work best in end-devices: • Distinctive ringing

  47. Programming SIP CGI • Follows Web-CGI. Unlike Web-CGI, SIP-CGI supports proxying and processes responses as well. • Language-indpendent (Perl, C, ...) • Communicates through input/output and environment variables. • CGI programs unlimited in their power. Drawback: Buggy scripts may affect server easily.

  48. Programming SIP Call Processing Language (CPL) • Special-purpose call processing language. • May be used by both SIP and H.323 servers. • Target scenario: users determine call processing logic executed at a server. • Limited languages scope makes sure server’s security will not get compromised. • Portability allows users to move CPL scripts across servers. • Scripts may be manually written, generated using convenient GUI tools, supplied by 3rd parties, ...

  49. Call Processing Language (CPL)Example

  50. Programming SIP Java Servlets • Compromise between security and power: still a powerful generic language but security provided by Java “sand-box”. • JAIN thought to be a generic API applicable to almost any signaling (SIP, H.323, PSTN, etc.)

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