1. BAI513 - PROTOCOLS RTP - RTCP BAIST – Network Management

2. Real-Time Transport Protocols RTP –End-to-end network transport functions for real-time data –Suitable framework for any encoding Complete Protocol : RTP framework + payload format –Used with RTCP optionally RTCP –Monitoring of data delivery –Minimal control and identification functionality

3. Introduction Services –Payload type identification –Sequence numbering –Timestamping –Delivery monitoring RTP over UDP/IP –Multiplexing –Checksum –Multicasting RTP does not provide –Timely delivery –Guaranteed QoS delivery –Out-of-order delivery prevention

4. RTP – A Transport Protocol

5. Real-Time Traffic Not all multimedia data is real time Stored Video Server Video Client Entire video transferred to client After receiving complete video, client begins playback Video Client Video Conference Server Individual video frames transferred to client as they occur Client plays back each video frame as it arrives There are tight constraints There are no tight constraints

6. Definitions RTP Payload –data transported by RTP in a packet RTP Packet –fixed RTP header + (list of contributing sources) + payload data Transport address –(network address, port) e.g) IP address + UDP port –need not to have a well-known port(default is 5004) RTP session –association among a set of participants communicating with RTP –defined by a pair of transport address

7. Definitions(cont’d) Synchronization Source(SSRC) –source of a stream of RTP packets –identified by a 32-bit ID(SSRC) in RTP header –globally unique in a RTP session –SSRC ID is randomly chosen –all packets from a SSRC form a part of the same timing & seq. # space –receiver groups packets by SSRC to playback Contributing Source(CCRC) –a source of a stream of RTP packets that has contributed to the combined stream produced by RTP mixer –Mixer inserts a list of the SSRC ids (CSRC list)

8. Mixer & Translator A Mixer is an intermediate system receives RTP packets from one or more sources possibly changes the data format combines the packets in some manner forwards a new RTP packet timing adjustment among several streams and generate its own timing –mixer becomes SSRC for the new combined packet Each source generates 64Kbit/s of Audio traffic SSRC : A SSRC : BSSRC : C Mixer E SSRC : E CSRC : A, B, C combines all three into a single 64Kbit/s stream

9. Translator A Translator is an intermediate system forwards RTP packets with their SSRC id intact e.g) –convert encoding without mixing –replicator from multicast to unicast –application-level filters in firewalls Frame Relay Each source generates 1Mbit/s of Video traffic converts each to a lower quality, 256Kbit/s stream SSRC : A SSRC : BSSRC : C SSRC : A SSRC : B SSRC : C

10. Example of Translator & Mixer E1 E2 M1 T1T2 E6 E7 M3 M2E3 E5 E4 E1 : 17 E2 : 1 M1 : 48 (1, 17) E4 : 47 M1 : 48 (1, 17) E4 : 47 E6 : 15 E3 : 64E5 : 45 M2 : 12 (64) M3 : 89 (64, 45) M1 : 48 (1, 17) E6 : 15 E4 : 47 M3 : 89 (64, 45) E : End system M : Mixer T : Translator source : SSRC (CSRCs)

11. Timing is Everything

12. Jj i tt tt er

13. Jitter Jitter is introduced in real-time data by the delay between packets It is the differing propagation delays of different packets A major issue in real time and streaming To prevent jitter, we can timestamp the packets and separate the arrival time from the playback time

14. Timestamping

15. Buffering Creates a queue of pending packets All delayed by some amount in order to compensate for maximum jitter Basically required for audio and video streams

16. Buffer Picture

17. Sequencing A sequence number on each packet is required for real-time traffic –Because packets may arrive out of order If subsequent packet already displayed, drop

18. Multicasting Real-time traffic needs the support of multicasting In order to scale to multiple senders and receivers Can’t use TCP, must use UDP –Also: TCP heavier weight –Also: can’t have retransmissions

19. Translation Translation means changing the encoding of a payload to a lower quality to match the bandwidth of the receiving network

20. RTP Header

21. Fields Version (2 bits)  2 P (1 bit)  if there is padding at the end X (1 bit)  if there is an extension header Contributor Count (4 bits)  how many M (1 bit)  generic marker Payload Type (7 bits)  audio, video, etc. Sequence number  16 bits

22. RTP Payload Types PTPayloadPTPayload 0 1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 PCMU audio 1016 audio G721 audio GSM audio Unassigned audio DV14 audio(8Khz) DV14 audio(16Khz) LPC audio PCMA audio G722 audio L16 audio(stereo) L16 audio(mono) TPS0 audio VSC audio MPA audio G728 audio 16-22 23 24 25 26 27 28 29 30 31 32 33 34-71 72-76 77-95 96-127 unassigned audio RGB8 video HDCC video CelB video JPEG video CUSM video nv video PicW video CPV video H261 video MPV video MP2T video unassigned video reserved unassigned dynamic

23. Fields Continued Timestamp  time differences between this and prior packet Synchronization Source  sender who synchronizes all other sources (mixer) Contributor Identifier(s)  others who are contributing source

24. RTP Data Transfer Protocol Multiplexing RTP sessions –by the destination transport address –In a teleconference The audio and video are transported in each separate session Profile specific modifications to the RTP header –The marker bit and payload type field has profile- specific info. –If particular application needs additional functionality independent of payload format define additional fixed fields to follow immediately after the SSRC field of header

25. RTP Header Extension if X bit is set –variable-length header extension is appended to the RTP header first 16-bits of header extension are left open for distinguishing ID or parameters for profile length counts the # of 32-bit words in the extension defined by profilelength Header extension 0 1531

26. RTCP Periodically transmitted to all participants of the session. Uses same distribution mechanism as RTP –such as multicasted UDP –RTP port # must be even number –RTCP port # must be odd (RTP + 1) Stackable RTCP messages

27. RTCP Feedback on the quality of the data distribution. Carries a persistent identifier (CNAME). Can count the number of session members. Provides minimum session control.

28. RTCP Message Types SR : Sender Report RR : Receiver Report SDES : Source DEScription BYE : End of participation APP : Application specific functions

29. SR length SSRC of packet sender RTP timestamp Sender’s packet count NTP timestamp VPRCPT=SR=200 Sender’s octet count SSRC_1 Cumulative number of packet lost Extend highest sequence number received Interarrival jitter Last SR (LSR) Delay since last SR(DLSR) Fraction lost SSRC_2... Profile-specific extensions

30. RR length SSRC of sender VPRCPT=RR=201 SSRC_1 Cumulative number of packet lost Extend highest sequence number received Interarrival jitter Last SR (LSR) Delay since last SR(DLSR) Fraction lost SSRC_2... Profile-specific extensions

31. SDES lengthVPSCPT=RR=202 SSRC/CSRC_1 SDES items SSRC/CSRC_2 SDES items

32. SDES Items CNAME : Canonical NAME –“user@host” or –“host” NAME : User name EMAIL : Email address PHONE : Phone number LOC : Geographical location

33. BYE length SSRC/CSRC VPSCPT=BYE=203... Reason for leavinglength

34. APP length SSRC/CSRC VPsubtypePT=APP=204 Name(ASCII) Application-dependent data

35. RTCP Transmission Interval For scalability, the interval must be increased as the group grows. Suggested control traffic fraction is 5%. Minimum interval is 5 seconds. (for the case of small group) The instant intervals are varied randomly to avoid synch. The more important, the more often.

36. RTCP Processing in Translators and Mixers If RTP data is modified, RTCP must be modified. –Aggregation of SDES –Modification of SR, RR It is advisable to combine RTCP packets from many sources to reduce overhead.

37. In Translators Without modification of data, without modification of RTCP With any transformation, it should reflect the characteristic & receiving quality.

38. SR sender info in Translators If data encoding is changed, sender’s byte count must be changed. If data packets are combined, sender’s packet count must be changed. If timestamp frequency is changed, RTP timestamp field must be changed. T RTP RTCP

39. SR/RR reception report in Translators Because of combination of several data packets, several fields must be changed. This may be complex. It can filter all reception reports, and generate new ones originating from itself. T RTP RTCP

40. SDES, BYE, APP in Translators SDES : Filters non-CNAME info if bandwidth is limited BYE, APP : Forwarded unchanged.

41. SR sender info in Mixer Does not forward at all. Generate new one of its own for both sides. M RTCP

42. SR/RR reception report info in Mixer Does not forward at all. Generate new one of its own for sender. M RTCP

43. Security Support in RTP Support confidentiality through encryption –use profile to specify security services and algorithms –DES in CBC mode as the default encryption algorithm May define additional payload types for encrypted encodings. Expect support for authentication and message integrity from lower layers.