1. Application Layer Protocols For 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. Everything runs over IP
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. A Real Life Scenario
Pure SIP based
In internet side

9. RTP = Real Time Protocol

10. Field Name in rtpTypes.h Content V version
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
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 Codec iLBC

13. iLBC Encoding
944 Figure 1.1

14. iLBC Decoding
944 Figure 1.2

15. iLBC residual coding from start state
944 figure 1.9

16. iLBC building an adaptive codebook from start state
944 figure 1.14

17. Codec response to packet loss - a waveform example

18. Codec response to packet loss - MOS test
Test conducted by Lockhead Martin (former Dynastat testlab)

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
Gerne eksempler fra side 10 I Søren og Steffens projekt

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

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

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

25. 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.

26. 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.

27. 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).

28. 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.

29. 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.

30. SIP Messages – Methods and Responses
SIP components communicate by exchanging SIP messages:
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.

31. SIP Headers SIP is an ASCII protocol
The SIP message is placed in a message Header
SIP Header
INVITE SIP/2.0
Via: SIP/2.0/UDP :5060
From:
To:
Call-ID:
CSeq: 100 INVITE
Expires: 180
User-Agent: Cisco IP Phone/ Rev. 1/ SIP enabled
Accept: application/sdp
Contact:
Content-Type: application/sdp

32. SIP Addressing
The SIP address is identified by a SIP URL, in the format:
Examples of SIP URLs:

33. 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.

34. 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.
Location/
Redirect Server
SIP Phone
User
Proxy/ Registration Server
REGISTER
REGISTER
200
200
SIP Messages:
REGISTER – Registers the address listed in the To header field.
200 – OK.

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

36. Resources WWW.IETF.ORG WWW.VOVIDA.ORG WWW.CS.COLUMBIA.EDU/SIP