Session Initiation Protocol (SIP) 陳 懷 恩 博 士 助 理 教 授 兼 計 算 機 中 心 資 訊 網 路 組 組 長 國 立 宜 蘭 大 學 資 工 系 TEL: # 340

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1 Session Initiation Protocol (SIP) 陳 懷 恩 博 士 助 理 教 授 兼 計 算 機 中 心 資 訊 網 路 組 組 長 國 立 宜 蘭 大 學 資 工 系 TEL: # 340

2 Outline Session Initiation Protocol SIP Extensions SIP Operation Examples References Q&A 2

3 Session Initiation Protocol

4 Introduction [1/2] SIP: Session Initiation Protocol Originally developed in the MMUSIC working group of the IETF Great interest led to a separate SIP working Group RFC 2543 RFC 3261 SIP is used in conjunction with several other IETF protocols, such as SDP and RTP. SIP has been selected as the interfaces between the IP Multimedia Core Network elements. RFC: Request for Comments MMUSIC: Multiparty Multimedia Session Control IETF: Internet Engineering Task Force SDP: Session Description Protocol RTP: Real-time Transport Protocol 4

5 Introduction [2/2] SIP is a powerful alternative to H.323. More flexible Simpler Easier to implement Better suited to the support of intelligent user devices Better suited to the implementation of advanced features SIP+MGCP/MEGACO will be the dominant VoIP signaling architecture of the future. MGCP/MEGACO: Media Gateway Control Protocol VoIP: Voice over IP 5

6 Architecture SIP is a signaling protocol. Setup a session Find user s current location Use SDP to carry session information Modify a session Teardown a session SIP signaling is separate from the media. SIP Messages RTP Streams SIP User Agent IP Network SIP User Agent 6

7 Network Entities [1/5] SIP User Agent (UA) User Agent Client (UAC) Application that sends requests The calling party User Agent Server (UAS) Application that receives requests and sends responses The called party 7

8 Network Entities [2/5] Location Server (Registrar) A database that maintains users location information Receive REGISTER requests Record who is at where now Support user mobility Generally combined with a proxy or redirect server 8

9 Network Entities [3/5] Proxy Server Receive requests Forward requests or send responses back to originator or both ex. 100 Trying Used for call forwarding, time-of-day routing or follow-me services 9

10 Network Entities [4/5] Redirect Server Receive requests Send responses Map the destination address to zero or more new addresses 302 Moved Temporarily 404 Not Found 10

11 Network Entities [5/5] 11

12 Registration Flow 12

13 Call Flow [1/3] UA to UA INVITE Transaction BYE Transaction 13

14 Call Flow [2/3] NotHere.org Through Proxy Server Location Server (2) Bob (3) UA (1) INVITE Proxy Server (4) INVITE UA (6) 200 OK (5) 200 OK (7) ACK (8) ACK (9) RTP Media Stream 14

15 Call Flow [3/3] NotHere.org Through Redirect Server Location Server UA (1) INVITE Proxy Server UA (4) 302 Moved temprarily Contact: (5) ACK (6) INVITE (7) 200 OK (8) ACK (9) RTP Media Stream 15

16 Message Syntax [1/2] Text-based Similar to HTTP More bandwidth consumption Two type Request message Response message 16

17 Message Syntax [2/2] Format Start-line Request: request-line Method SP Request-URI SP SIP-version CRLF Specify the type of request Response: status-line SIP-version SP Status-code SP Reason-phrase CRLF Indicate the success of failure of a given request Headers* Additional information of the request or response CRLF [Message body] Describe the type of session Examined only at the two end SDP 17

18 SIP Addressing URI: Uniform Resource Identifiers Format: ex. ex. ex. ex. 18

19 Request Messages [1/2] Method SP Request-URI SP SIP-version CRLF Methods REGISTER, INVITE, ACK, BYE, CANCEL, OPTIONS REGISTER Create a mapping of public address and current address Can register to multiple servers Can register several current addresses with one public address in one server INVITE Initiate a session Contains information of the calling and called parties inside Headers Contains the type of media to be used inside Message body ACK only when receiving the final response 19

20 Request Messages [2/2] ACK Ensure receiving the final response of INVITE request BYE Terminate a session Can be issued by either the calling or called party CANCEL Terminate a pending request OPTIONS Query one s capabilities Request-URI URI of the destination (UAS) SIP-version Generally SIP/2.0 20

21 Response Messages SIP-version SP Status-code SP Reason-phrase CRLF Status-code Three-digit number (i.e., 100 Trying or 200 OK) 1xx, Informational 2xx, Success 3xx, Redirection 4xx, Request Failure 5xx, Server Failure 6xx, Global Failure 1xx is considered provisional responses where others are considered final responses Reason-phrase A textual description of the outcome 21

22 Headers Provide additional information ex. From header indicates the UAC Four type General Headers Request Headers Response Headers Entity Headers 22

23 General Headers [1/2] Apply to both request and response messages Provide basic information ex. To URI of called party if there exists called party INVITE, OPTIONS ex. From URI of UAC ex. Call-ID Uniquely identifier of the session ex. Contact URI for future communication 23

24 General Headers [2/2] ex. Via Address and port information of UA or proxies in sequence in the routing path Prevent request looping Ensure that responses will route via the same path as requests Add when send requests Remove when send responses ex. CSeq Command sequence CSeq = CSeq: DIGIT SP Method Consecutive requests within a session use the same Call-ID but increasing sequence number (DIGIT) Requests and responses within a transaction use the same CSeq ex. INVITE 200 OK ACK ex. BYE 200 OK 24

25 Request Headers Apply only to request messages Provide additional information about the request ex. Subject Summary or the nature of the call ex. Max-Forwards Number of proxies or gateways that can forward the request downstream ex. Priority Urgency of the request 25

26 Response Headers Apply only to response messages Provide further information about the response that cannot be included in the status line ex. Retry-After Time in seconds that services or the calling party might being available ex. Unsupported Features that are not supported by the UAS if it is asked to support those 26

27 Entity Headers Define information about the message body ex. Content-Length Length of the message body ex. Content-Type Media type of the message body ex. application/sdp ex. Context-Encoding Information about encoding and decoding the message body ex. Context-Language 27

28 Registration Message Flow [1/2] [M1] Register

29 Registration Message Flow [2/2] [M1] Register [M2] 200 OK

30 Call Message Flow [1/15] [M01] INVITE

31 Call Message Flow [2/15] [M01] INVITE [M02] INVITE

32 Call Message Flow [3/15] [M01] INVITE [M03] 100 Trying [M02] INVITE

33 Call Message Flow [4/15] [M01] INVITE [M03] 100 Trying [M02] INVITE [M04] 100 Trying [M04] SIP/ Trying Via: SIP/2.0/UDP ;branch=z9hG4bK944c.70b Via: SIP/2.0/UDP :12264 From: To: Call-ID: CSeq: 1 INVITE User-Agent: Windows RTC/1.0 Content-Length: 0 33

34 Call Message Flow [5/15] [M01] INVITE [M03] 100 Trying [M02] INVITE [M04] 100 Trying [M05] 180 Ringing

35 Call Message Flow [6/15] [M01] INVITE [M03] 100 Trying [M06] 180 Ringing [M02] INVITE [M04] 100 Trying [M05] 180 Ringing [M06] SIP/ Ringing Via: SIP/2.0/UDP :12264 From: To: Call-ID: CSeq: 1 INVITE Record-Route: User-Agent: Windows RTC/1.0 Content-Length: 0 35

36 Call Message Flow [7/15] [M01] INVITE [M03] 100 Trying [M06] 180 Ringing [M02] INVITE [M04] 100 Trying [M05] 180 Ringing [M07] 200 OK [M07-1] SIP/ OK Via: SIP/2.0/UDP ;branch=z9hG4bK944c.70b Via: SIP/2.0/UDP :12264 From: To: Call-ID: CSeq: 1 INVITE Record-Route: Contact: <sip: :15450> User-Agent: Windows RTC/1.0 Content-Type: application/sdp Content-Length: 455 [M07-2] v=0 o=r100-lin 0 0 IN IP s=session c=in IP b=ct:1000 t=0 0 m=audio RTP/AVP a=rtpmap:97 red/

37 Call Message Flow [8/15] [M01] INVITE [M03] 100 Trying [M06] 180 Ringing [M08] 200 OK [M02] INVITE [M04] 100 Trying [M05] 180 Ringing [M07] 200 OK

38 Call Message Flow [9/15] [M01] INVITE [M03] 100 Trying [M06] 180 Ringing [M08] 200 OK [M02] INVITE [M04] 100 Trying [M05] 180 Ringing [M07] 200 OK [M09] ACK

39 Call Message Flow [10/15] [M01] INVITE [M03] 100 Trying [M06] 180 Ringing [M08] 200 OK [M02] INVITE [M04] 100 Trying [M05] 180 Ringing [M07] 200 OK [M09] ACK [M10] ACK [M10] ACK sip: :15450 SIP/2.0 Record-Route: Via: SIP/2.0/UDP ;branch=0 Via: SIP/2.0/UDP :12264 Max-Forwards: 69 From: To: Call-ID: CSeq: 1 ACK User-Agent: RTC/1.2 Content-Length: 0 39

40 Call Message Flow [11/15] [M01] INVITE [M03] 100 Trying [M06] 180 Ringing [M08] 200 OK [M02] INVITE [M04] 100 Trying [M05] 180 Ringing [M07] 200 OK [M09] ACK [M11] RTP Media [M10] ACK 40

41 Call Message Flow [12/15] [M01] INVITE [M03] 100 Trying [M06] 180 Ringing [M08] 200 OK [M02] INVITE [M04] 100 Trying [M05] 180 Ringing [M07] 200 OK [M09] ACK [M12] BYE [M11] RTP Media [M10] ACK 41

42 Call Message Flow [13/15] [M01] INVITE [M03] 100 Trying [M06] 180 Ringing [M08] 200 OK [M02] INVITE [M04] 100 Trying [M05] 180 Ringing [M07] 200 OK [M09] ACK [M12] BYE [M11] RTP Media [M10] ACK [M13] BYE [M13] BYE sip: :15450 SIP/2.0 Record-Route: Via: SIP/2.0/UDP ;branch=z9hG4bK644c.887cba62.0 Via: SIP/2.0/UDP :12264 Max-Forwards: 69 From: To: Call-ID: CSeq: 2 BYE User-Agent: RTC/1.2 Content-Length: 0 42

43 Call Message Flow [14/15] [M01] INVITE [M03] 100 Trying [M06] 180 Ringing [M08] 200 OK [M02] INVITE [M04] 100 Trying [M05] 180 Ringing [M07] 200 OK [M09] ACK [M12] BYE [M11] RTP Media [M10] ACK [M13] BYE [M14] 200 OK [M14] SIP/ OK Via: SIP/2.0/UDP ;branch=z9hG4bK644c.887cba62.0 Via: SIP/2.0/UDP :12264 From: To: Call-ID: CSeq: 2 BYE User-Agent: Windows RTC/1.0 Content-Length: 0 43

44 Call Message Flow [15/15] [M01] INVITE [M03] 100 Trying [M06] 180 Ringing [M08] 200 OK [M02] INVITE [M04] 100 Trying [M05] 180 Ringing [M07] 200 OK [M09] ACK [M12] BYE [M15] 200 OK [M11] RTP Media [M10] ACK [M13] BYE [M14] 200 OK [M15] SIP/ OK Via: SIP/2.0/UDP :12264 From: To: Call-ID: CSeq: 2 BYE User-Agent: Windows RTC/1.0 Content-Length: 0 44

45 SDP Introduction SDP: Session Description Protocol RFC 2327 Text-based Describing the media to be exchanged between the parties The structure of SDP Session level info. Media level info. 45

46 SDP Syntax Contains a number of lines of text Each line is field=value field is exactly one character (case-sensitive) value may be one or more components Two main blocks Session level fields Media level fields Begin with media description field (m=) Some field can be applied at both session and media levels. The value applied at the media level overrides that at the session level. 46

47 SDP Fields [1/3] v=(protocol version) ex. v=0 o=(session owner or creator) username, session ID, version, network type, address type, address ex. o=r100-lin 0 0 IN IP s=(session name) ex. s=session c=(connection information) network type, address type, connection address ex. c=in IP

48 SDP Fields [2/3] b=(bandwidth information) In kilobits per second ex. b=ct:1000 t=(time of the session) For pre-arranged multi-party conference start time, stop time ex. t=0 0 m=(media description) media type, port, transport, media payload format list ex. m=audio RTP/AVP a=(attribute) Several formats Session level: information about the whole session Media level: information about the media stream Take rtpmap attribute as example: rtpmap, payload type, encoding name, clock rate ex. a=rtpmap:97 red/

49 SDP Fields [3/3] Ordering of fields Media level m=(media description) i=(media information) c=(connection information) b=(bandwidth information) k=(encryption key) a=(attributes) Session level v=(protocol version) o=(session owner or creator) s=(session name) i=(session information) u=(uri) e=( address) p=(phone number) c=(connection information) b=(bandwidth information) t=(time description) r=(repeat information) z=(time zone adjustments) k=(encryption key) a=(attributes) 49

50 Using SDP in SIP [1/3] SIP uses SDP in an offer/answer mode. An agreement between the two parties as to the types of media they are willing to share INVITE <-> 200 OK UAC (1) INVITE I can use both XXX and YYY as media type UAS (2) 200 OK No problem. Use XXX 50

51 Using SDP in SIP [2/3] 200 OK <-> ACK Special Case!! Unsupported should also be returned with a port number of zero. ex. m=audio 0 RTP/AVP 2 51

52 Using SDP in SIP [3/3] A mismatch happens if can t support all codec from the other part. 488 Not Acceptable Here 606 Not Acceptable Caller issues a new INVITE request 52

53 One number service [1/2] One number service is also called follow me service. A s public address is mapped to several current addresses and a proxy can find more than one of them. If B calls A through the proxy, this proxy can send an INVITE to a number of locations at the same time. This type of parallel search is known as forking. 53

54 One number service [2/2] Alice Proxy Server (1) REGISTER Bob (2) 200 OK (3) REGISTER Bob (5) INVITE Bob (6) 100 Trying (4) 200 OK (7) INVITE Bob (8) INVITE Bob (9) 200 OK (11) 200 OK (for INVITE) (13) ACK (10) CANCEL (12) 200 OK (for CANCEL) (14) ACK (15) Media Stream 54

55 SIP Extensions

56 SIP Extensions Because of great simplicity, flexibility and other features, SIP has attracted enormous interest. Traditional telecommunications companies Cable TV providers ISPs A large number of extensions to SIP have been proposed. For presence and instant message For communication with SS7 PSTN network For phone transfer 56

57 INFO Method RFC 2976 INFO method is defined for transferring information during an ongoing session. DTMF digits Account balance information Pre-paid service 57

58 SUBSCRIBE & NOTIFY Method RFC 3265 Some SIP-based application may need to know the information of a certain event of other users. presence SUBSCRIBE Subscribe the information of a certain event of one user Subscribed event in Event header NOTIFY Send current or updated information of a certain event to who subscribed me 58

59 MESSAGE Method RFC 3428 SIMPLE (SIP for Instant Messaging and Presence Leveraging Extensions) is a separate IETF working group. MESSAGE Context of message is in the message body. A MESSAGE request does not establish a SIP dialog. 59

60 REFER Method RFC 3515 The sender of the request can instruct the receiver to contact a third party. Call transfer REFER The URI of the third party is in Refer-to header. The dialog between Alice and Bob remains established. Bob must inform Alice about the status of the referece. In message body Content-type: message/sipfrag 60

61 UPDATE Method RFC 3311 UPDATE allows a client to update parameters of a session but has no impact on the state of a dialog. Change the codec 61

62 PRACK Method&RSeq, Rack Headers [1/2] Lost of provisional response (1xx) may cause problems when interoperating with other network. 180, 183 Q.931 alerting or ISUP ACM To drive a state machine RFC 3262 defines PRACK Method and some Headers to support reliability of provisional response messages. Supported: 100rel RSeq header Response sequence +1 when retransmit RAck header Response ACK In PRACK RSeq+CSeq PRACK Method This mechanism should not apply to 100 Trying and the default timer value is 0.5 second. 62

63 PRACK Method&RSeq, Rack Headers [2/2] Alice (1) INVITE Supported: 100rel CSeq: 1 INVITE Bomb!! (2) 180 Ringing Require: 100rel RSeq: Cseq: 1 INVITE (3) 180 Ringing Require: 100rel RSeq: Cseq: 1 INVITE Bob (4) PRACK RAck: INVITE CSeq: 2 PRACK (2) 200 OK CSeq: 2 PRACK 63

64 SIP UA Operation Examples -Windows Messenger Operation -NTP VoIP Platform

65 NTP VoIP Platform in NCTU SIP Proxy IPtel SER SIP PSTN GW CISCO 2600 PSTN Network SIP SIP SIP PSTN GW SIP Megaco SIP UA Call Server Media GW 65

66 Numbering Plan Example 號 碼 目 的 地 說 明 [XXXXX] Media Gateway 交 大 校 內 分 機, 由 Media GW 下 車 035[XXXXXX] Media Gateway 新 竹 市 內 電 話, 由 Media GW 下 車 02[XXXXXXXX] sip.ntu.edu.tw 台 北 長 途 電 話, 轉 交 台 大 SIP Server 0939[XXXXXX] CISCO 2600 大 哥 大, 由 CISCO 下 車 0938[XXXXXX] CISCO 2600 大 哥 大, 由 CISCO 下 車 [XXX] sip.ipv6.club.tw 網 路 電 話, 交 給 交 大 SIP Server 66

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