1. Session Initiation Protocol (SIP) Ram Dantu (Compiled from different sources, see the references list)

2. Internet Telephony 2 SIP based VoIP Architecture Legacy PBX SIP User Agents (UA) Application Services eMail LDAP Oracle XML SIP RTP (Media) SIP CPL 3pcc PSTN CAS or PRI INTELLIGENTSERVICESINTELLIGENTSERVICES SIP Proxy, Registrar & Redirect Servers

3. Internet Telephony 3 Basic SIP Call-Flow SIP UA1 SIP UA2 INVITE w/ SDP for Media Negotiation 100 Trying 180/183 Ringing w/ SDP for Media Negotiation 200 OK BYE MEDIA ACK

4. Internet Telephony 4 SIP Call Flow with Proxy Server Proxy Server Register OK (200) Register OK (200) Invite Trying (100) Ringing (180) OK (200) ACK RTP/RTCP media channels

5. VoIP Migration

6. Internet Telephony 6 Customer Premises IP Core Network Step1: IPPBX deployments in Enterprises DNS Server for URL resolution - Large enterprises will handle VOIP calls directly - PSTN connectivity provided by Media Gateways - Regulation can not stop spammers outside USA (similar to SMTP spam) Customer Premises PSTNNetwork

7. Internet Telephony 7 Customer Premises Carrier Network Softswitches, MGW VoIP Proxy Server, SGW SGC, VoIP Centrex Server, Internet STEP 2: Hosted IP Centrex FW, NAT, VoIP service provided by Carrier Networks

8. Internet Telephony 8 Customer Premises Carrier Network Step 3: Carrier VoIP Network Softswitches, MGW VoIP Proxy Server, SGW SGC, VoIP Centrix Server, Internet - VoIP FW, NAT and Security provided by Carriers VoIP Trunk

9. SIP Architecture

10. Internet Telephony 10 The Popularity of SIP Originally Developed in the MMUSIC A separate SIP working group RFC 2543, RFC 3261 Many developers SIP + MGCP/MEGACO The VoIP signaling in the future “ back-off ” or SIPit (SIP Interoperability Tests) Test products against each other Will be hosted by ETSI

11. Internet Telephony 11 SIP Architecture A signaling protocol The setup, modification, and tear-down of multimedia sessions SIP + SDP Describe the session characteristics Separate signaling and media streams

12. Internet Telephony 12 SIP Network Entities Clients User agent clients Application programs sending SIP requests Servers Responds to clients ’ requests Clients and servers may be in the same platform Proxy Acts as both clients and servers

13. Internet Telephony 13 Four types of servers Proxy servers Handle requests or forward requests to other servers Can be used for call forwarding

14. Internet Telephony 14 Redirect servers Map the destination address to zero or more new addresses Do not initiate any SIP requests

15. Internet Telephony 15 A user agent server Accept SIP requests and contacts the user The user responds → an SIP response A SIP device E.g., an SIP-enabled telephone A registrar Accepts SIP REGISTER requests Indicating the user is at a particular address Typically combined with a proxy or redirect server

16. Internet Telephony 16 SIP Call Establishment It is simple A number of interim responses

17. Internet Telephony 17 SIP Advantages Attempt to keep the signaling as simple as possible Offer a great deal of flexibility Various pieces of information can be included within the messages Including non-standard information Enable the users to make intelligent decisions The user has control of call handling No need to subscribe call features

18. Internet Telephony 18 Call Completion to Busy Subscriber service

19. Internet Telephony 19 Via contains the address (e.g., pc33.atlanta.com) Contact contains a SIP or SIPS URI that represents a direct route to contact the called party, usually composed of username at a fuly qualified domain name (FQDN). While the FQDN is preferred, many end systems do not have registered domain names, so IP addresses are permitted. While Via header field tells other elements where to send response, the Contact header field tells other elements where the called party can be reached directly. In a response, Via, To, From, Call-ID, and CSeq header fields are copied from the INVITE request. In addition to DNS and location service lookups, proxy servers can make flexible “ routing decisions ” to decide where to send a request. For example, if Bob ’ s SIP phone returned 486 (busy) response, the biloxi.com proxy server could proxy the INVITE to Bob ’ s voicemail server. A proxy server can also send an INVITE to a number of locations at the same time. This type of parallel search is known as forking.

20. Internet Telephony 20 After learning the end point addresses, the end points can communicate directly

21. Internet Telephony 21 Overview of SIP Messaging Syntax Text-based Similar to HTTP SIP messages message = start-line *message-header CRLF [message-body] start-line = request-line | status-line Request-line specifies the type of request The response line The success or failure of a given request

22. Internet Telephony 22 Message headers Additional information of the request or response E.g., The originator and recipient Retry-after header Subject header Message body Describe the type of session The media format SDP, Session Description Protocol Could include an ISDN User Part message Examined only at the two ends

23. Internet Telephony 23 SIP Requests method SP request-URI SP SIP-version CRLF request-URI The address of the destination Methods INVITE, ACK, OPTIONS, BYE, CANCLE, REGISTER extensions: INFO, REFER, UPDATE, … INVITE Initiate a session Information of the calling and called parties The type of media ～ IAM (initial address message) of ISUP ACK only the final response

24. Internet Telephony 24 BYE Terminate a session Can be issued by either the calling or called party Options Query a server as to its capabilities A particular type of media The response if sent an INVITE CANCEL Terminate a pending request E.g., an INVITE did not receive a final response

25. Internet Telephony 25 REGISTER Log in and register the address with a SIP server “ all SIP servers ” – multicast address (224.0.1.1750) Can register with multiple servers Can have several registrations with one server INFO RFC 2976 Transfer information during an ongoing session DTMF digits account balance information midcall signaling information generated in another network

26. Internet Telephony 26 SIP Responses SIP version SP status code SP reason-phrase CRLF reason-phrase A textual description of the outcome Could be presented to the user status code A three-digit number 1XX Informational 2XX Success (only code 200 is defined) 3XX Redirection 4XX Request Failure 5XX Server Failure 6XX Global Failure All responses, except for 1XX, are considered final Should be ACKed

27. Internet Telephony 27 “ One number ” service

28. Internet Telephony 28 SIP Addressing SIP URLs (Uniform Resource Locators) user@host E.g., sip:collins@home.net sip:3344556789@telco.net Supplement the URL sip:3344556789@telco.net;user=phone sip:user:password@host:port;uri-parameters?headers

29. Internet Telephony 29 Message Headers Provide further information about the message ～ information elements E.g., To:header in an INVITE The called party From:header The caling party Four main categories General, request, response, and entity headers A list in Table 5-2 Mapping in Table 5-3

30. Internet Telephony 30 General Headers Used in both requests and responses Basic information E.g., To:, From:, Call-ID:, … Contact: A URL for future communication May be different from the From: header Requests passed through proxies

31. Internet Telephony 31 Request Headers Apply only to SIP requests Addition information about the request or the client E.g., Subject: Priority:, urgency of the request Authorization:, authentication of the request originator Response Headers Further information about the response E.g., Unsupported:, features Retry-After

32. Internet Telephony 32 Entity Header Session information presented to the user Session description, SDP The RTP payload type, an address and port Content-Length, the length of the message body Content-Type, the media type of the message Content-Encoding, for message compression Content Disposition, Content-Language, Allow, used in a Request to indicate the set of methods supported Expires, the date and time

33. Internet Telephony 33 Example of SIP Message Sequences Registration Via: Call-ID: host-specific Content-Length: Zero, no msg body Cseg: Avoid ambiguity Expires: TTL 0, unreg Contact: *

34. Internet Telephony 34 Invitation A two-party call Subject: optional Content-Type: application/sdp

35. Internet Telephony 35

36. Internet Telephony 36 Termination of a Call Cseq: Has changed

37. Internet Telephony 37 Redirect Servers An alternative address 302, Moved temporarily Another INVITE Same Call-ID Cseq ++

38. Internet Telephony 38 Proxy Servers Entity headers are omitted Changes the Req-URI Via: The path Loop detected, 482 For a response The 1 st Via: header Checked removed

39. Internet Telephony 39

40. Internet Telephony 40 Proxy state Can be either stateless or stateful Record-Route: The messages and responses may not pass through the same proxy Use Contact: A Proxy might require that it remains in the signaling path In particular, for a stateful proxy Insert its address into the Record-Route: header The response includes the Record-Route: header The Record-Route: header is used in the subsequent requests The Route: header = the Record-Route: header in reverse order, excluding the first proxy Each proxy remove the next from the Route: header

41. Internet Telephony 41 Forking Proxy “ fork ” requests A user is registered at several locations ;branch=xxx

42. Internet Telephony 42

43. Internet Telephony 43 The Session Description Protocol The message body SDP, RFC 2327 The Structure of SDP Session Level Info Name The originator The time Media Level Info Media type Port number Transport protocol Media format

44. Internet Telephony 44 SDP session description structure

45. Internet Telephony 45 SDP Syntax A number of lines of text In each line field=value Session-level fields first Media-level fields Begin with media description field (m=)

46. Internet Telephony 46 Mandatory Fields v=(protocol version) o=(session origin or creator and session id) s=(session name), a text string t=(time of the session) t= NTP time values in seconds m=(media) m= Media type The transport port The transport protocol The media format, an RTP payload format

47. Internet Telephony 47 Optional Fileds i=(session information) A text description At both session and media levels u=(URI of description) Where further session information can be obtained Only at session level e=(e-mail address) Who is responsible for the session Only at the session level p=(phone number) Only at the session level

48. Internet Telephony 48 c=(connection information) Connection type, network type, and connection address At session or media level b=(bandwidth information) In kilobits per second At session or media level r= For regularly scheduled session How often and how many times

49. Internet Telephony 49 z=(timezone adjustments) z=.... For regularly scheduled session Standard time and Daylight Savings Time k=(encryption key) k= : An encryption key or a mechanism to obtain it At session or media level a=(attributes) Describe additional attributes

50. Internet Telephony 50 Ordering of Fields Session Level Protocol version (v) Origin (o) Session name (s) Session information (i) URI (u) E-mail address (e) Phone number (p) Connection info (c) Bandwidth info (b) Time description (t) Repeat info (r) Time zone adjustments (z) Encryption key (k) Attributes (a) Media level Media description (m) Media info (i) Connection info (c) Optional if specified at the session level Bandwidth info (b) Encryption key (k) Attributes (a)

51. Internet Telephony 51 Subfields Field = … Origin (o) Username, the originator ’ s login id or “ - ” session ID A unique ID Make use of NTP timestamp version, a version number for this particular session network type A text string; IN refers to Internet address type IP4, IP6 Address, a fully-qualified domain name or the IP address o=mhandley 2890844526 2890842807 IN IP4 126.16.64.4

52. Internet Telephony 52 Connection Data The network and address at which media data are to be received Network type, address type, connection address c=IN IP4 224.2.17.12/127 Media Information Media type Audio, video, application, data, or control Port, 1024-65535 Format List the various types of media RTP/AVP payload types m= audio 45678 RTP/AVP 15 3 0 G.728, GSM, G.711

53. Internet Telephony 53 Attributes Property attribute a=sendonly a=recvonly value attribute a=orient:landscape rtpmap attribute The use of dynamic payload type a=rtpmap: / [/ ]. m=video 54678 RTP/AVP 98 a=rtpmap 98 L16/16000/2

54. Internet Telephony 54 Usage of SDP with SIP SIP for the establishment of multimedia sessions SDP – a structured language for describing the sessions The entity header

55. Internet Telephony 55 Negotiation of Media Fig 5-15 G.728 is selected If a mismatch 488 or 606 Not Acceptable A Warning header INVITE with multiple media streams Unsupported should also be returned With a port number of zero

56. Internet Telephony 56

57. Internet Telephony 57 Offer/answer

58. Internet Telephony 58

59. Internet Telephony 59 OPTIONS Method Determine the capabilities of a potential called party