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Application Layer Protocols For Real-Time Media Transmission

Application Layer Protocols For Real-Time Media Transmission. The Layers – Again (OSI model). Application/ Representation/Session. Application/ Representation/Session. Transport. Transport. Network. Network. Network. Link/ Physical. Link/ Physical. Link/ Physical. Link/ Physical.

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Application Layer Protocols For Real-Time Media Transmission

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  1. Application Layer ProtocolsFor Real-Time Media Transmission

  2. The Layers – Again(OSI model) Application/ Representation/Session Application/ Representation/Session Transport Transport Network Network Network Link/ Physical Link/ Physical Link/ Physical Link/ Physical

  3. The Layers – Again(Internet model) Application Application Transport Transport Internetwork Internetwork Internetwork Subnetwork Subnetwork Subnetwork Subnetwork

  4. The Layers(Web browsing example) HTTP HTTP TCP TCP IP IP routing IP Ethernet Ethernet Ethernet Ethernet

  5. The Layers(Real Time Media) SIP/RTP/RTCP SIP/RTP/RTCP TCP/UDP TCP/UDP IP IP routing IP Ethernet Ethernet Ethernet Ethernet

  6. The Role of IP Everything runs over IP IP IP runs over Everything

  7. Multi-Multi-Multi-Multi • Subnetwork: multiple communications share the same physical channel • Internetwork: Each communication may pass multiple subnetworks • Transport: End-To-End abstraction • Application: multiple users take part in same communication session using multiple types of media

  8. Pure SIP based In internet side A Real Life Scenario

  9. RTP = Real Time Protocol

  10. Field Name in rtpTypes.h Content V version Version of the RTP. Always equal to two. P padding Not used. Padding may be needed by some encryption algorithms with a required fixed block size. X extension Not used. Used if the header is followed by a header extension. CSRC count (CC) count Number of CSRC identifiers that follow the fixed header M marker For applications which send no packets during silence, the first packet of a talk spurt (first packet after a silence period) is distinguished by setting the marker bit in the RTP data header. Applications without silence suppression set the bit to zero. Payload type type Identifies the format of the RTP payload at any given time. A profile specifies a default static mapping of payload type codes to payload formats. Additional payload codes may be defined dynamically through SIP negotiation. Audio applications operating under the AV profile specified in RFC 1890 should, at minimum, be able to send and receive payload types 0 (G.711U) and 5 (DVI4). This allows interoperability without format negotiation and successful negotiation with a conference control protocol. The payload types currently defined in this profile carry either audio or video, but not both. Sequence number sequence The sequence number increments by one for each RTP data packet sent, and may be used by the receiver to detect packet loss and to restore packet sequence. The initial value of the sequence number is random (unpredictable) to make known-plaintext attacks on encryption more difficult

  11. Field Name in rtpTypes.h Content Timestamp timestamp The timestamp reflects the sampling instant of the first octet in the RTP data packet. If an audio application reads blocks covering 160 sampling periods from the input device, the timestamp would be increased by 160 for each such block, regardless of whether the block is transmitted in a packet or dropped as silent. The timing information enables the application to synchronize audio and video data. SSRC ssrc The SSRC field identifies the synchronization source. A receiver groups packets by synchronization source for playback. Examples of synchronization sources include the sender of a stream of packets derived from a signal source such as a microphone or a camera. This identifier is chosen randomly, with the intent that no two synchronization sources within the same RTP session will have the same SSRC identifier. CSRC list startOfCsrc 0 to 15 items The CSRC list is used when a payload is composed of data from several sources and serves to identify these sources. The number of identifiers is given by the CC field. If there are more than 15 contributing sources, only 15 may be identified. CSRC identifiers are inserted by mixers, using the SSRC identifiers of contributing sources. For example, for audio packets the SSRC identifiers of all sources that were mixed together to create a packet are listed, allowing correct talker indication at the receiver.

  12. An Example CodeciLBC

  13. iLBC Encoding

  14. iLBC Decoding

  15. iLBC residual coding from start state

  16. iLBC building an adaptive codebook from start state

  17. Codec response to packet loss - a waveform example

  18. Codec response to packet loss- MOS test

  19. Codec response to packet loss- some signals • G729 at 0% packet loss • iLBC at 0% packet loss • G729 at 15% packet loss • iLBC at 15% packet loss

  20. Structure of the specification • A draft on the codec draft-ietf-avt-ilbc-codec-04.txt • A draft on the RTP payload profile draft-ietf-avt-rtp-ilbc-04.txt

  21. PLC: Packet Loss Concealment iLBC PLC with normal muting iLBC PLC with rulebased muting

  22. Adaptive Protection At this point: present adaptive protection results from the Norcom Talk

  23. SIP = Session Initiation Protocol = An application layer signaling protocol that defines initiation, modification and termination of interactive, multimedia communication sessions between users.

  24. SIP Framework Instant Messaging • Session initiation. • Multiple users. • Interactive multimedia applications. Personal Mobility Voice Calls Conferencing Email Distance Learning MPEG, MP3, Audio, HTML,XML Video Conferencing

  25. Redirect Server Registrar Server Location Server Proxy Server Proxy Server SIP Distributed Architecture SIP Components PSTN User Agent Gateway

  26. User Agents • An application that initiates, receives and terminates calls. • User Agent Clients (UAC) – An entity that initiates a call. • User Agent Server (UAS) – An entity that receives a call. • Both UAC and UAS can terminate a call.

  27. Proxy Server • An intermediary program that acts as both a server and a client to make requests on behalf of other clients. • Requests are serviced internally or by passing them on, possibly after translation, to other servers. • Interprets, rewrites or translates a request message before forwarding it.

  28. Location Server • A location server is used by a SIP redirect or proxy server to obtain information about a called party’s possible location(s).

  29. Redirect Server • A server that accepts a SIP request, maps the address into zero or more new addresses and returns these addresses to the client. • Unlike a proxy server, the redirect server does not initiate its own SIP request. • Unlike a user agent server, the redirect server does not accept or terminate calls.

  30. Registrar Server • A server that accepts REGISTER requests. • The register server may support authentication. • A registrar server is typically co-located with a proxy or redirect server and may offer location services.

  31. SIP Methods: INVITE – Initiates a call by inviting user to participate in session. ACK - Confirms that the client has received a final response to an INVITE request. BYE - Indicates termination of the call. CANCEL - Cancels a pending request. REGISTER – Registers the user agent. OPTIONS – Used to query the capabilities of a server. INFO – Used to carry out-of-bound information, such as DTMF digits. SIP Responses: 1xx - Informational Messages. 2xx - Successful Responses. 3xx - Redirection Responses. 4xx - Request Failure Responses. 5xx - Server Failure Responses. 6xx - Global Failures Responses. SIP Messages – Methods and Responses SIP components communicate by exchanging SIP messages:

  32. SIP Headers • SIP is an ASCII protocol • The SIP message is placed in a message Header • -------------------------------------------------- SIP Header ----------------------------------------------------------------- INVITE sip:5120@192.168.36.180 SIP/2.0 Via: SIP/2.0/UDP 192.168.6.21:5060 From: sip:5121@192.168.6.21 To: <sip:5120@192.168.36.180> Call-ID: c2943000-e0563-2a1ce-2e323931@192.168.6.21 CSeq: 100 INVITE Expires: 180 User-Agent: Cisco IP Phone/ Rev. 1/ SIP enabled Accept: application/sdp Contact: sip:5121@192.168.6.21:5060 Content-Type: application/sdp

  33. SIP Addressing • The SIP address is identified by a SIP URL, in the format: user@host. • Examples of SIP URLs: • sip:hostname@vovida.org • sip:hostname@192.168.10.1 • sip:14083831088@vovida.org

  34. Process for Establishing Communication • Establishing communication using SIP usually occurs in six steps: • Registering, initiating and locating the user. • Determine the media to use – involves delivering a description of the session that the user is invited to. • Determine the willingness of the called party to communicate – the called party must send a response message to indicate willingness to communicate – accept or reject. • Call setup. • Call modification or handling – example, call transfer (optional). • Call termination.

  35. Location/ Redirect Server REGISTER REGISTER 200 200 SIP Phone User Registration • Each time a user turns on the SIP user client (SIP IP Phone, PC, or other SIP device), the client registers with the proxy/registration server. • Registration can also occur when the SIP user client needs to inform the proxy/registration server of its location. • The registration information is periodically refreshed and each user client must re-register with the proxy/registration server. • Typically the proxy/registration server will forward this information to be saved in the location/redirect server. Proxy/ Registration Server SIP Messages: REGISTER – Registers the address listed in the To header field. 200 – OK.

  36. Proxy Server Location/Redirect Server Proxy Server INVITE 180 (Ringing) 180 (Ringing) 200 (OK) 200 (OK) 200 (OK) 200 (OK) INVITE INVITE 180 (Ringing) ACK ACK BYE BYE 200 (OK) 200 (OK) RTP MEDIA PATH INVITE ACK BYE User Agent User Agent Simplified SIP Call Setup and Teardown 302 (Moved Temporarily) ACK INVITE Call Setup 302 (Moved Temporarily) ACK MediaPath Call Teardown

  37. Resources • WWW.IETF.ORG • WWW.VOVIDA.ORG • WWW.CS.COLUMBIA.EDU/SIP • WWW.IPTEL.ORG • WWW.SIPFOUNDRY.ORG

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