The basics of multimedia exchange. VoIP and MultiMedia /50

Transcription

1 The basics of multimedia exchange VoIP and MultiMedia /50

2 Where did it all start? VoIP and MultiMedia /50

3 The basics of IP telephony. network core (registrars, proxies, ) Bob Address: B Port: Pb Alice Address: A Port: Pa VoIP and MultiMedia 2012 emil.ivov@jitsi.org 3/50

4 The basics of IP telephony. network core (registrars, proxies, ) Bob Address: B Port: Pb Alice Address: A Port: Pa VoIP and MultiMedia 2012 emil.ivov@jitsi.org 4/50

5 The basics of IP telephony. network core (registrars, proxies, ) MEDIA Bob Address: B Port: Pb Alice Address: A Port: Pa VoIP and MultiMedia 2012 emil.ivov@jitsi.org 5/50

6 Intra Domain SIP Signaling 4 thomas@u-strasbg.fr 5 SIP Soft Client 3 7 emcho@u-strasbg.fr 6 2 u-strasbg.fr Proxy Server 1 SIP Phone u-strasbg.fr Registrar and Location Service 1. Call Thomas - INVITE 2. Query Where is Thomas@u-strasbg.fr? (non-sip) 3. Response (non-sip) 4. Proxied Call - INVITE 5. Response - OK 6. Response - OK 7. Multimedia Chanel Establised RTP Streams VoIP and MultiMedia 2012 emil.ivov@jitsi.org 6/50

7 Protocol Details SIP URIs VoIP and MultiMedia /50

8 Example SIP URIs SIP URI with username: SIP URI without a username: sip:example.com sip:x.example.com sip: SIP URI with parameters: sip:abc@example.com;transport=tcp;user=phone IPv6 SIP URI: sip:emcho@[fe80::5445:5245:444f]:5560 VoIP and MultiMedia 2012 emil.ivov@jitsi.org 8/50

9 Protocol Details Message Structure VoIP and MultiMedia /50

10 Example SIP Messages INVITE SIP/2.0 Via: SIP/2.0/UDP ;branch=1 From: To: Contact: Call-ID: Cseq: INVITE SIP/ Not Found Via: SIP/2.0/UDP ; branch=1 From: To: Call-ID: Cseq: INVITE v=0 o=user IP s=. t=0 0 c=in IP m=audio RTP/AVP 0 a=rtpmap:0 PCMU/8000 VoIP and MultiMedia 2012 emil.ivov@jitsi.org 10/50

11 Message Structure: First Line The first line, determines the semantical type of the message: Request Response Request line contains: method: determines the type of the request SIP URI: determines the destination of the request SIP protocol version <METHOD> <Request-URI> SIP/2.0! Response line contains SIP protocol version status-code: digital response code reason phrase SIP/2.0 <Status-Code> <Reason-Phrase>! VoIP and MultiMedia /50

12 The Session Initiation Protocol Methods are the verbs of the protocol Original six methods in version 2.0 of SIP INVITE REGISTER BYE ACK CANCEL OPTIONS Both Requests and Responses can carry SIP bodies usually SDP, but could be a JPEG or JAVA script SIP Responses carry a status code and a reason phrase - human readable 180 Ringing 402 Payment Required Request Format Request line Several Headers Empty Line Message Body Response Format Status line Several Headers Empty Line Message Body VoIP and MultiMedia 2012 emil.ivov@jitsi.org 12/50

13 SIP Requests and Responses 6 Methods in the core specification - RFC3261 COMMAND FUNCTION INVITE - Initiate Call ACK - Confirm Final Response BYE - Terminate an established media session CANCEL - Cancel Searches & Ringing OPTIONS - Features support by other side REGISTER -Register with location server SIP Response Codes 1xx informational responses are not retransmitted if lost * 2xx success responses are delivered with end to reliability 3xx - 6xx non-successful responses delivered hop-by-hop CODE RANGE RESPONSE CLASS EXAMPLES 1XX Informational Provisional -Queued, Ringing, Being Forwarded 2XX Success Final -OK, Accepted 3XX Redirection Final -Moved Temporarily, Moved Permanently 4XX Client error Final -Payment Required, Method Not Allowed 5XX Server error Final -Not Implemented, Service Unavailable 6XX Global failure Final -Busy Everywhere, Decline VoIP and MultiMedia 2012 emil.ivov@jitsi.org 13/50 13

14 Error Codes Note:Informational text is for the user and any number of 1xx messages can be sent e.g 100 Message received 100 Looking up number 100 Found carrier "AT&T 100 Dialling number Note: Many codes are same as HTTP SIP specific codes start x80 Informational 100 Trying 180 Ringing 181 Call Is Being Forwarded 182 Queued 183 Session Progress Success 200 OK 202 Accepted Redirection 300 Multiple Choices 301 Moved Permanently 302 Moved Temporarily 303 See Other 305 Use Proxy 380 Alternative Service Client error 400 Bad Request 401 Unauthorized 402 Payment Required 403 Forbidden 404 Not Found 405 Method Not Allowed 406 Not Acceptable 407 Proxy Authentication Required 408 Request Timeout 409 Conflict 410 Gone 411 Length Required 413 Request Entity Too Large 414 Request-URI Too Large 415 Unsupported Media Type 420 Bad Extension 480 Temporarily not available 481 Call Leg/Transaction Does Not Exist 482 Loop Detected 483 Too Many Hops 484 Address Incomplete 485 Ambiguous 486 Busy Here 487 Request Cancelled 488 Not Acceptable Here Server error 500 Internal Server Error 501 Not Implemented 502 Bad Gateway 503 Service Unavailable 504 Gateway Time-out 505 SIP Version not supported Global failure 600 Busy Everywhere 603 Decline 604 Does not exist anywhere 606 Not Acceptable VoIP and MultiMedia /50 14

15 SIP Headers The format of a request message: <METHOD> <Request-URI> SIP/2.0 CRLF <Header1>: <Value1> CRLF <Header2>: <Value2> CRLF <HeaderN>: <ValueN> CRLF CRLF <Message Body> The format of a response message: SIP/2.0 <Status-Code> <Reason-Phrase> CRLF <Header1>: <Value1> CRLF <Header2>: <Value2> CRLF <HeaderN>: <ValueN> CRLF CRLF <Message Body> VoIP and MultiMedia 2012 emil.ivov@jitsi.org 15/50

16 SIP Headers Examples of Headers Used in Requests and Responses HEADER FUNCTION Call-ID -Used to uniquely identify a call between two user agents Contact -Used to convey URL of original resource requested or request originator CSeq -Command Sequence identifies out of sequence requests & retransmissions From -Identifies originator of request To -Indicates recipient of request Subject -Optional header indicating subject of media session Content-Length -Number of octets in the message body Content-Type -Indicates Internet media type. If not present application/sdp is assumed User-Agent -Provides additional information about the user agent e.g. manufacturer Server -Provides additional information about the User Agent Server Via -Records the route taken by a request and used to route response Record-Route -Used to force all requests between UAs to be routed through a Proxy Route -Forces routing through a path extracted from a Record-Route header Authorization -Carries credentials of user agent to a server Encryption -Used to specify the portion of a SIP message that has been encrypted Hide -Requests next hop proxy to encrypt the Via headers Priority -Allow the user agent to set the priority of a request: e.g. urgent, emergency Supported -List one more options implemented in a user agent or server Unsupported -Indicates features that are not supported by the server VoIP and MultiMedia 2012 emil.ivov@jitsi.org 16/50

17 Example SIP Request INVITE SIP/2.0 Via: SIP/2.0/UDP ;branch=1 From: To: Contact: Call-ID: Cseq: INVITE Content-Type: application/sdp Content-Length: 274 v=0 o=user IP s=. t=0 0 c=in IP m=audio RTP/AVP 0 a=rtpmap:0 PCMU/8000 VoIP and MultiMedia 2012 emil.ivov@jitsi.org 17/50

18 Example SIP Request Request line INVITE SIP/2.0 Headers Via: SIP/2.0/UDP ;branch=1 From: To: Contact: Call-ID: Cseq: INVITE Content-Type: application/sdp Content-Length: 274 Empty line Body (if any) v=0 o=user IP s=. t=0 0 c=in IP m=audio RTP/AVP 0 a=rtpmap:0 PCMU/8000 VoIP and MultiMedia 2012 emil.ivov@jitsi.org 18/50

19 Example SIP Response SIP/ Not Found Via: SIP/2.0/UDP ; branch=1 From: To: Call-ID: Cseq: INVITE VoIP and MultiMedia /50

20 Example SIP Response Response line SIP/ Not Found Headers Via: SIP/2.0/UDP ; branch=1 From: To: Call-ID: Cseq: INVITE Empty line Body (if any) VoIP and MultiMedia /50

21 A Very Basic SIP Call Flow Alice Bob Transaction INVITE 100 Trying 180 Ringing 200 OK ACK MEDIA Dialog BYE OK VoIP and MultiMedia /50

22 Registrars and Registrations LittleGuy Registrar REGISTER To: LittleGuy Contact: 200 OK Contact: VoIP and MultiMedia /50

23 An example REGISTER request REGISTER sip:b.com SIP/2.0 Via: SIP/2.0/UDP From: To: Contact: Expires: 3600 Call-ID: CSeq: 1 REGISTER VoIP and MultiMedia 2012 emil.ivov@jitsi.org 23/50

24 An example REGISTER request Request-URI registration domain Who s registering AOR Contact REGISTER sip:b.com SIP/2.0 Via: SIP/2.0/UDP From: sip:barbara@b.com To: sip:barbara@b.com Contact: <sip:barbara@ > Duration in seconds Expires: 3600 Empty line Call-ID: @ CSeq: 1 REGISTER VoIP and MultiMedia 2012 emil.ivov@jitsi.org 24/50

25 An example REGISTER response SIP/ Ok Via: SIP/2.0/UDP From: To: es=3600 Contact:<sip: >;expires=345 Contact:<sip: >;expires=1000 Call-ID: CSeq: REGISTER VoIP and MultiMedia /50

26 An example REGISTER response SIP/ Ok Via: SIP/2.0/UDP Who s regisering AOR List of all Contact headers for know AORs From: sip:barbara@b.com;tag= To: sip:barbara@b.com;tag=jjf223 Contact:<sip:barbara@ >;expir es=3600 Contact:<sip: >;expires=345 Contact:<sip: >;expires=1000 Call-ID: @ CSeq: REGISTER Empty line VoIP and MultiMedia 2012 emil.ivov@jitsi.org 26/50

27 REGISTER: refresh, cancel, query It is up to the user agent to refresh registrations of Contact addresses. In order to do so, a UA has to resend its innitial REGISTER request. In order to cancel a Contact registration, a user agent has to set its Expires time to zero To: sip:barbara@b.com Contact: <sip:barbara@ > Expires: 0 In order to cancel all contact address of records, a UA could use an asterisk (*) To: sip:barbara@b.com Contact: * Expires: 0 Omitting the Contact header would not modify any AOR and the corresponding response would contain all existin AORs. VoIP and MultiMedia 2012 emil.ivov@jitsi.org 27/50

28 Authentication and Authorization SIP provides a stateless, challenge-based mechanism for authentication that is based on authentication in HTTP. Any time that a proxy server or UA receives a request, it MAY challenge the initiator of the request to provide assurance of its identity. Digest - Most common authentication mechanism (provides message authentication and replay protection but not message integrity or confidentiality). VoIP and MultiMedia 2012 emil.ivov@jitsi.org 28/50

29 Challenge and response LittleGuy REGISTER WWW-Authenticate: Digest realm="biloxi.com", qop="auth,auth-int", nonce="dcd98b7102dd2f0e8b11d0f600bfb0c093", opaque="5ccc069c403ebaf9f0171e9517f40e Unauthorized REGISTER 200 OK Authorization: Digest username="bob", realm="biloxi.com", nonce="dcd98b7102dd2f0e8b11d0f600bfb0c093", qop=auth, nc= , cnonce="0a4f113b", response="6629fae49393a c4ef1, opaque="5ccc069c403ebaf9f0171e9517f40e41" VoIP and MultiMedia /50

30 Registrars and Registrations LittleGuy Registrar REGISTER From: LittleGuy Contact: 401 Unauthorized Contact: WWW-Authenticate: <Authentication Challenge>! REGISTER From: LittleGuy Contact: Authorization: <Authentication Response>! 200 OK Contact: VoIP and MultiMedia /50

31 Inter Domain SIP Signaling sip.jitsi.org voip.unistra.fr Transaction INVITE 1 INVITE 2 TRYING 3 INVITE 4 TRYING 5 RINGING 6 RINGING 8 RINGING 7 OK 9 OK 10 OK 11 ACK MEDIA Dialog BYE OK VoIP and MultiMedia 2012 emil.ivov@jitsi.org 31/50

32 By the way,. do you know what SRV records are? Locating SIP Servers RFC ~$ host -t SRV _sip._udp.jitsi.org _sip._udp.jitsi.org SRV sip3.jitsi.org.! _sip._udp.jitsi.org SRV sip1.jitsi.org.! _sip._udp.jitsi.org SRV sip2.jitsi.org.! _sip._udp.jitsi.org SRV sip4.jitsi.org.! _sip._udp.jitsi.org SRV sip5.jitsi.org.! ~$ host -t A sip3.jitsi.org sip3.jitsi.org has address ~$ host -t AAAA sip3.jitsi.org sip3.jitsi.org has AAAA address 2001:660:4701:1001::1! VoIP and MultiMedia 2012 emil.ivov@jitsi.org 32/50

33 SIP Transactions - Client Server Relationship Client Transaction Server Transaction Client Transaction Server Transaction Client Transaction Server Transaction UAC Outbound Proxy Inbound Proxy UAS VoIP and MultiMedia 2012 emil.ivov@jitsi.org 33/50

34 Invite Client Transactions Timer A fires Reset A Invite Sent Calling INVITE from TU INVITE Sent Timer B fires or Transport Error Inform TU ACK Sent Response to TU 1xx 1xx to TU ACK Sent 1xx 1xx to TU Proceeding Completed ACK Sent Resp. to TU Timer D fires 2xx 2xx to TU 2xx 2xx to TU Transport Error Inform TU Terminated VoIP and MultiMedia 2012 emil.ivov@jitsi.org 34/50

35 Invite Server Transactions INVITE Send Response Proceeding INVITE Pass INV to TU send 100 if TU won't in 200ms from TU Send Response Transport err. Inform TU INVITE Send Response from TU Send Response ACK - Completed Confirmed 2xx fromtu Send response Timer G fires Send response Timer H fires Or transp. Err inform TU Timer I fires - Terminated VoIP and MultiMedia 2012 emil.ivov@jitsi.org 35/50

36 Examples Examples of call establishment. VoIP and MultiMedia /50

37 Sample Invite Transaction - Busy Alice Proxy1 Proxy2 Bob INVITE Trying3 486 Busy 10 ACK 11 INVITE Trying5 486 Busy 8 ACK 9 INVITE Busy 6 ACK 7 VoIP and MultiMedia 2012 emil.ivov@jitsi.org 37/50

38 Sample Invite Transaction - No Response Alice Proxy1 Proxy2 Bob INVITE Trying3 480 NR 14 ACK 15 INVITE Trying5 480 NR 12 ACK 13 INVITE 4 INVITE 6 INVITE 7 INVITE 8 INVITE 9 INVITE 10 INVITE 11 VoIP and MultiMedia 2012 emil.ivov@jitsi.org 38/50

39 Sample Invite Transaction - Temporarily Unavailable Alice Proxy1 Proxy2 Bob INVITE Trying3 180 Ringing 8 INVITE Trying5 180 Ringing 7 INVITE Ringing NA 13 ACK NA 11 ACK NA 9 ACK 10 VoIP and MultiMedia 2012 emil.ivov@jitsi.org 39/50

40 Sample Invite Transaction - No Answer (CANCEL) Alice Proxy1 Proxy2 Bob INVITE Trying3 180 Ringing 8 CANCEL OK RT 19 ACK 20 INVITE Trying5 180 Ringing 7 CANCEL OK RT 17 ACK 18 INVITE Ringing 6 CANCEL OK RT 15 ACK 16 VoIP and MultiMedia 2012 emil.ivov@jitsi.org 40/50

41 INVITE transaction - Success despite a failing proxy!!! Alice Proxy1 Proxy2 Bob Alice Proxy1 Proxy2 Bob INVITE 1 INVITE 2 INVITE 3 ACK 18 ACK 19 INVITE 4 INVITE 5 INVITE 6 MEDIA INVITE 7 INVITE 8 BYE 21 BYE Auth. Req 9 ACK OK OK 23 INVITE Trying Ringing OK 17 INVITE Ringing OK 16 VoIP and MultiMedia 2012 emil.ivov@jitsi.org 41/50

42 Non-Invite Client Transactions Timer E fires Reset E Request resent Trying Request from TU Request Sent Timer F fires or Transport Error Inform TU Response to TU Timer E fires Resend request Resp. to TU 1xx 1xx to TU Proceeding Completed Timer K fires 1xx 1xx to TU Timer F Or trans. err inform TU Terminated VoIP and MultiMedia 2012 emil.ivov@jitsi.org 42/50

43 Non-Invite Server Transactions Trying Request Received Pass Req to TU from TU Send Response Transport Error Inform TU Request Send Response Proceeding 1xx fromtu Send response 1xx fromtu Send response Request Send Response Transport Error Inform TU Completed Timer J fires - Terminated from TU Send Response Timer H fires Or transp. Err inform TU VoIP and MultiMedia 2012 emil.ivov@jitsi.org 43/50

44 Examples Examples of non-invite transactions VoIP and MultiMedia /50

45 Sample non-invite transaction LittleGuy Registrar REGISTER 401 Unauthorized REGISTER REGISTER STOP 200 OK VoIP and MultiMedia /50

46 Dialogs Dialogs are a key concept for a user agent. A dialog represents a peer-to-peer SIP relationship between two user agents that persists for some time The dialog represents a context in which to interpret SIP messages. A dialog is identified at each UA with a dialog ID, which consists of a Call- ID value, a local tag and a remote tag. From: Bob <sips:bob@biloxi.example.com>;tag=a73kszlfl! To: Alice <sips:alice@biloxi.example.com>;tag= ! Call-ID: 1j9FpLxk3uxtm8tn@biloxi.example.com! VoIP and MultiMedia 2012 emil.ivov@jitsi.org 46/50

47 Dialogs (Continued) Dialogs are created through the generation of non-failure responses to requests with specific methods (RFC 3261 only specifies one such request - INVITE) For INVITE initiated dialogs, terminating the dialog consists of sending a BYE request. Extensions specifying other dialog initiating requests, also define the way they are ended. VoIP and MultiMedia 2012 emil.ivov@jitsi.org 47/50

48 A Dialog Use Case (reinvite) PROXY INVITE Trying RINGING OK INVITE RINGING OK ACK MEDIA Bob changes his IP Address reinvite OK ACK AUDIO SESSION BYE OK VoIP and MultiMedia /50

49 The basics of multimedia exchange VoIP and MultiMedia /50