Multimedia Communication

Transcription

1 Multimedia Communication Session Description Protocol SDP Session Announcement Protocol SAP Realtime Streaming Protocol RTSP Session Initiation Protocol - SIP Dr. Andreas Kassler Slide 1

2 SDP Slide 2

3 SDP SDP Session Description Protocol IETF RFC 2327 Originally designed for multicast conference announcements Can now be used for unicast, multicast, Describes multimedia sessions (stream and contents) Text based form Use ISO character set in UTF-8 encoding Format, timing, and authorship of streamed media For session announcement, session invitation and session setup Can be used together with many other protocols SAP, SIP, RTSP, MIME, HTTP,. SDPng is XML based and will replace SDP Slide 3

4 SDP Session Description Session Level Information Protocol Version Originator, Session ID Session Name Session Title Media Description Media 1 Media Name, Media Transport Connection Information Media Description Media 2 Media Name, Media Transport Connection Information Slide 4

5 SDP SDP Media Descriptions SDP Media Descriptions Attributes preceding media descriptions apply to all otherwise apply to nearest media descriptions Attributes as subfields of media description type, e.g. audio video transport, e.g. RTP H.320 media format, e.g. H.261 G.711 codec specific attributes Slide 5

6 SDP Session level Description Optional (marked with*) and mandatory fields Exact ordering required (according to table) v= v= o= o= s= s= i=* i=* u=* u=* e=* e=* p=* p=* c=* c=* b=* b=* (Protocol version, in in this this case case 0) 0) (owner/creator and and session identifier) (Session Name) Name) (Session Information) (URI (URI of of additional information) (owner address) (owner phone phone number) (connection information - - not not required if if included in in all all media) (bandwidth information) o =<username> <session id> <version> <network type> <address type> <address> c=<network type> <address type> <connection address> Slide 6

7 SDP Session level Description Time identifiers specify e.g. if session is repeated every day at 9 p.m. t= t= r=* r=* z=* z=* k=* k=* a=* a=* (time (time the the session is is active) (zero (zero or or more more repeat times) (time (time zone zone adjustments) (encryption key) key) (zero (zero or or more more session attribute lines) t=<start time> <stop time> a=<attribute>:<value> Slide 7

8 Session level Description Media descriptions SDP m= m= i=* i=* c=* c=* b=* b=* k=* k=* a=* a=* (media name and and transport address) (media title) title) (connection information - - optional if if included at at session-level) (bandwidth information) (encryption key) key) (zero (zero or or more more media attribute lines) c=<network type> <address type> <connection address> m=<media> <port> <transport> <fmt list> b=ct:128 //total bandwidth for conference b=as:64 //application specific, for one app k=prompt //ask user for key a=rtpmap:97 L16/11025/2 ; a=sendonly; a=recvonly; a=orient:landscape Slide 8

9 SDP Example v=0 v=0 o=mhandley o=mhandley IN IN IP4 IP s=sdp s=sdp Seminar Seminar i=a i=a Seminar Seminar on on the the session session description description protocol protocol u= u= (Mark (Mark Handley) Handley) c=in c=in IP4 IP / /127 t= t= a=recvonly a=recvonly m=audio m=audio RTP/AVP RTP/AVP 0 0 m=video RTP/AVP 31 m=video RTP/AVP 31 m=audio m=audio RTP/AVP RTP/AVP a=rtpmap:96 a=rtpmap:96 L8/8000 L8/8000 a=rtpmap:97 a=rtpmap:97 L16/11025/2 L16/11025/2 m=application m=application udp udp wb wb a=orient:portrait a=orient:portrait Session Session Description Description Audio Audio Media Media Description Description Video Video Media Media Description Description Audio Audio Media Media Description Description (dynamic (dynamic PT) PT) Application Application Media Media Description Description Slide 9

10 SAP Slide 10

11 SAP SAP Session Announcement Protocol IETF RFC 2974 Advertises Conference Announcements by session originator to well defined multicast group sent on port 9875, IP TTL should be 255 to receive announcements, simply join the multicast group Contents of SAP messages are SDP payload Low announcement rate (every few minutes) total bandwidth on single SAP group below 4kbit/s Session deletion explicit timeout (SDP carries end-time of session) implicit timeout delete session from session cache if session announcement is not received for ten times announcement period explicit deletion special SAP packet Slide 11

12 RTSP Slide 12

13 RTSP RTSP Realtime Streaming Protocol IETF RFC 2326 control over multiple data delivery sessions of data with realtime properties RTSP acts like a remote control Supports ondemand Scenarios Media retrieval from server Invitation of media server to a conference Recording a conference Slide 13

14 RTSP RTSP Realtime Streaming Protocol Text based Independent of transport protocol, e.g. can use UDP or TCP normally transported over TCP, as commands need reliable transfer, in-order delivery performance problems in wireless Independent of Session Description e.g. can use SDP, XML,.. Similar to HTTP, but Client server AND Server client Server maintains session state Media data out of band, using e.g. RTP Slide 14

15 RTSP Presentations RTSP Multimedia presentation identified by URLs protocol: rtsp hostname: server that contains the presentation port: identifies the port where RTSP messages should be sent to Examples rtsp://media.disney.com:554/terminator rtsp://media.disney.com:554/terminator/audiotrack rtsp://media.disney.com:554/terminator/videotrack Slide 15

16 RTSP Session Concept Idea client RTSP requests to start a presentation by a media server uses RTSP Server responds with session identifier identifies state in server and client no recovery from system crash Slide 16

17 RTSP RTSP Messages and Format RTSP similar to HTTP v1.1 application level object oriented message format Request/Response Type General header Request/Response Header Entity header Message body (if necessary) Slide 17

18 RTSP Message details Request Type command, followed by URL Response Type RTSP numeric code plus textual description (e.g. 200 ok ) General header e.g. via, connection, date Request Header e.g. acceptable encodings, software being used,.. Response header e.g. age of response, type of server, Entity header optional info about message body, like length, type, expires Message Body data associated with request Slide 18

19 RTSP Requests SETUP RTSP Server allocates resources and starts RTSP session PLAY Server starts media streaming PAUSE Temporarily stop a stream TEARDOWN Free stream resources, destroy RTSP session Slide 19

20 RTSP Requests OPTIONS RTSP What methods are supported? Server sends list ANNOUNCE Register description of media object using SDP DESCRIBE Get low level media description server sends SDP RECORD Server starts recording the media stream REDIRECT Redirects the client to a new server SET_PARAMETER, GET_PARAMETER Additional parameter to be exchanged, e.g. for Device or encoding control, camera angle, quantization Slide 20

21 Example WebServer AudioServer 1. Get Mediadescription (SDP) Client 2. OpenStreams (using RTSP) 3. PLAY 4. TEARDOWN VideoServer Slide 21

22 Example Can also use DESCRIBE to media server Receive session ID provide URL, sessionid, start and stop time can pause at any time or change playback by QoS specifying Group new time Client WebServer MediaServers HTTP GET SDP Presentation Description RTSP SETUP 200 OK RTSP PLAY 200 OK RTP Audio and Video RTCP RTSP TEARDOWN 200 OK allocate state and resources deallocate state and resources Slide 22

23 GET /twister.sdp HTTP/1.1 Host: Accept: application/sdp Client RTSP/ OK 200 OK CSeq: 3 RTSP/ OK RTP Audio and Video CSeq: 2 RTCP Session: Range: smpte=0:10:00-0:20:00 RTSP TEARDOWN QoS RTP-Info: Group url=rtsp://audio.example.com/twister/audio.en; seq=876655;rtptime= OK HTTP/ OK Content-Type: application/sdp v=0 o= IN IP Example SETUP rtsp://audio.example.com/twister/audio.en RTSP/1.0 CSeq: 1 s=rtsp Session m=audio 0 RTP/AVP 0 a=control:rtsp://audio.example.com/twister/audio.en m=video 0 RTP/AVP WebServer 31 MediaServers a=control:rtsp://video.example.com/twister/video HTTP GET SDP Presentation Description Transport: RTP/AVP/UDP;unicast;client_port= RTSP/ OK CSeq: 1 Session: RTSP SETUP Transport: RTP/AVP/UDP;unicast;client_port= ; server_port= OK PLAY rtsp://audio.example.com/twister/audio.en RTSP/1.0 CSeq: 2 Session: Range: smpte=0:10:00- RTSP PLAY TEARDOWN rtsp://audio.example.com/twister/audio.en RTSP/1.0 CSeq: 3 Session: Slide 23

24 SIP Slide 24

25 SIP SIP Session Initiation Protocol IETF RFC RFC 2543, 2976, 3087, , 3311,3312 Initiate, Control and Teardown telephony services over Internet E-2-E call signalling Session Establishment User s current location? Common capabilities and preferences? Carries opaque session descriptions Sessions Modification Session Teardown Properties distributed light weight, based on UDP peer-to-peer human readable Slide 25

26 SIP Features Flexibility Use URL as addresses Users can move to new locations access features from anywhere Users can define responses to call setup requests, e.g. redirect to voice recorder, Functionality Setting up point-to-point or multiparty multimedia conferences Call handling capabilities are extensive Negotiation of media capabilities is possible Interleaving of resource management is possible during call setup Slide 26

27 SIP Components User Agent Client (UAC) Sends, Forwards SIP Requests In Endsystems and Proxies User Agent Server (UAS) Listens for Call requests Determines Response Changes SIP state Forward to user Executes Program User Agent (UA) = UAC + UAS Slide 27

28 SIP Components Registrar Domain Proxy+Registry Placed in the Network Receives Registrations Current user location Capabilities, QoS of user? Redirect Server Registrar + Location Service Placed in the Network Redirects calls to another Server Slide 28

29 SIP Components Proxy Server Used for forwarding calls and finding participants Only Intra-Domain Never replies with 100 Trying stateless current state of the call stored at endpoints simplifies design scalability Proxy Server (Stateful) Proxy Server (Stateful) Proxy Request to another server can fork request to multiple servers Interdomain Proxy Keeps track of call state For AAAC, Service/Transport Domain, Slide 29

30 SIP SIP Messages message format SIP-Message := Request Response Generic-message := Start_line * Message_header CRLF[body] Start_line := Request_line Status_line Request_line contains methods like INVITE Message_header := ( general-header request-header response-header entity-header ) Slide 30

31 SIP REGISTER Method Purpose: register users current location multicast to all SIP servers multicast addr. sip.mcast.net ( ) Can register at multiple locations contact all one at a time until successful response contact all at once, wait for first response, ignore rest redirect server sends all locations to caller, caller decides order of contact Slide 31

32 SIP INVITE Method Purpose: invite callee to join a session initiates the call change call state, e.g. put someone on hold Message includes caller identification FROM callee identification TO CSEQ proposed SDP of call parameters (optional) Media negotiation response contains type of possible/acceptable media descriptions caller re-invites with the common denominator proxy information (VIA), only if call was routed through proxies Slide 32

33 SIP BYE Method Purpose: terminate call or call request terminate media flow Not necessary to wait for response SIP Headers SIP Headers Allowed categories of information that can be contained within SIP message ~ 40 headers defined, more info in RFCs Slide 33

34 SIP Call routing Call routing INVITE is sent through one or more proxies each proxy adds VIA line to header contains ID of the proxy Callees response can then be routed along the reverse path proxies can be notified all messages between caller and callee proxies remove their via line in the response state is encapsulated in the SIP message itself Slide 34

35 SIP Response codes Response codes Call Status: Trying, for proxies 180 Ringing 181 Call Is Being Forwarded 182 Queued Success: OK Slide 35

36 SIP Response codes Response codes Redirection: Ambiguous address returns possible matches 301 Moved Permanently returns new address(ess) to try 302 Moved Temporarily returns temporarily address for call forwarding 305 Use proxy Slide 36

37 SIP Response codes Response codes Client Error: Syntax Error 401 Unauthorized 402 Payment Required 403 Forbidden Request 404 Called user not found 405 Method not allowed 408 Server cannot respond within the given time constraints 409 conflict between requests (e.g. duplicate registration) 410 gone, forwarding address not available 420 server does not understand extension mechanism 480 temporarily not available Slide 37

38 SIP Response codes Client Error: transaction does not exist 482 loop detected 483 too many hops 485 ambiguous 486 busy or unwilling to accept, may indicate when to try again Slide 38

39 SIP Response codes Response codes Server Error: internal error 501 not implemented 502 bad gateway 503 service temporarily not available 504 gateway timeout 505 SIP version not supported Global failure: callee busy everywhere 603 callee declined call 604 callee does not exist anywhere 606 media choice not acceptable Slide 39

40 USER A USER B INVITE 180 Ringing 200 OK ACK BYE 200 OK Normal Call Flow Normal Call Flow Normal Call Flow SIP Mandatory fields Mandatory fields every message contains call ID call sequence number CSEQ: identifies transaction in call signalling FROM TO VIA only if proxies are involved Slide 40

41 SIP Response Jon initiates call to by issuing INVITE message Media 0 SIP Client sip:eve@isi.edu Request QoS Proxies Group may directly reply to caller, caller 1 DNS 2 3 Lookup SRV Record eve@sipgw.isi.edu Sip.isi.edu SIP Proxy SIP Client (User Agent Server) 5 eve Slide 41 Distributed Systems Department Proxy of isi.edu? Proxy is sip.isi.edu SIP SIP Example Relay Mode Proxies Proxies are are optional, users users can can call call directly, directly, if if they they know know IP-@ IP-@ to to use use Proxies may directly reply to caller, caller then then contacts callee callee directly directly redirect redirect mode mode send callers invitation to sip.isi.edu Location Service 7 6 Proxy is sipgw.cs.isi.edu Sipgw.cs.isi.edu 8 SIP Proxy -knows how to reach eve -DB is updated whenever eve registers new location 9

42 SIP Call Flow Details Call Flow Details Call Flow Details USER A PROXY PROXY USER B INVITE 407 Proxy Authenticate ACK ACK as answer to all responses >1xx INVITE INVITE 100 Trying INVITE 100 Trying 180 Ringing 180 Ringing 180 Ringing 200 OK 200 OK 200 OK ACK ACK ACK BOTH WAY RTP Media Data BYE BYE BYE 200 OK 200 OK 200 OK Hangs up Slide 42

43 SIP SIP Example Redirect Mode Redirect Server 1: INVITE 0 3: ACK 2: MOVED TEMPORARILY SIP Client 4: INVITE Wants to contact 5: 200 OK Request Response Media SIP Client (User Agent Server) Slide 43

44 Try all locations at once Try all locations at once Try all locations at once USER at addr 1 USER at addr 2 REGISTER (addr1) INVITE 200 OK ACK SIP Registrar/ Proxy REGISTER (addr2) 200 OK 200 OK INVITE CANCEL 200 OK (CANCEL) BOTH WAY RTP Media Data Caller INVITE TRYING 200 OK (INVITE) ACK Slide 44

45 SIP Caller Callee INVITE 486 BUSY, try at 10 pm ACK INVITE 180 RINGING 200 OK ACK BOTH WAY RTP Media Data BYE 200 OK Implement Services e.g. automatic ringback at e.g. automatic ringback at a time time specified by by the the callee Slide 45

46 SIP SIP does NOT provide QoS Control SIP uses Preconditions Can be used to signal successful resource reservations Resources must be reserved before the phone rings Invitations might indicate via SDP that QoS is mandatory Call setup should proceed only when resources have been reserved Define preconditions Define direction of resource reservation e.g. sendrcv SIP extension method UPDATE Updates state of session Can be used for resource reservation states Slide 46

47 SIP USER A USER B INVITE (SDP1) 183 Session Progress (SDP2) PRACK Resource Reservation 200 OK (PRACK) Resource Reservation UPDATE (SDP3) 200 OK (UPDATE) (SDP4) 180 Ringing PRACK 200 OK (PRACK) 200 OK (INVITE) ACK QoS preconditions in SDP1: direction =sendrcv B tells A that it reserves resources from B to A but A must reserve from A to B B does not inform user as both way reservation is required A indicates to B that resources are reserved B tells A that all preconditions are met and notifies user Normal session establishment continues Slide 47