1 / 41

RTP/RTCP/RTSP

Real-Time Protocols. RTP/RTCP/RTSP. Amit Hetawal University of Delaware CISC 856 -Fall 2005. Thanks to Professor Amer. Overview. History of streaming media Streaming performance requirements Protocol stack for multimedia services Real-time transport protocol (RTP)

eman
Download Presentation

RTP/RTCP/RTSP

An Image/Link below is provided (as is) to download presentation Download Policy: Content on the Website is provided to you AS IS for your information and personal use and may not be sold / licensed / shared on other websites without getting consent from its author. Content is provided to you AS IS for your information and personal use only. Download presentation by click this link. While downloading, if for some reason you are not able to download a presentation, the publisher may have deleted the file from their server. During download, if you can't get a presentation, the file might be deleted by the publisher.

E N D

Presentation Transcript

  1. Real-Time Protocols RTP/RTCP/RTSP Amit Hetawal University of Delaware CISC 856 -Fall 2005 Thanks to Professor Amer

  2. Overview History of streaming media Streaming performance requirements Protocol stack for multimedia services Real-time transport protocol (RTP) RTP control protocol (RTCP) Real-time streaming protocol (RTSP)

  3. Brief history of streaming media

  4. Real-time multimedia streaming • Real-time multimedia applications • Video teleconferencing • Internet Telephony (VoIP) • Internet audio, video streaming (A-PDUs)

  5. Streaming performance requirements • Sequencing – to report PDU loss • to report PDU reordering • to perform out-of-order decoding • Time stamping and Buffering • for play out • for jitterand delay calculation • Payload type identification • for media interpretation • Error concealment –covers up errors from lost PDU by using redundancy in most-adjacent-frame • Quality of Service (QoS) feedback – from receiver to sender for operation adjustment • Rate control –sender reduces sending rate adaptively to network congestion

  6. Ideal Timing – no jitter 30 seconds 00.00.00 00.00.10 First RTP-PDU application 00.00.11 00.00.20 Second RTP-PDU 00.00.21 00.00.30 Third RTP-PDU 00.00.31 Send time Play time

  7. Reality – jitter delay 00.00.00 00.00.10 First RTP-PDU 00.00.11 00.00.20 Second RTP-PDU 00.00.21 00.00.30 00.00.25 Third RTP-PDU 00.00.40 00.00.35 00.00.37 Fourth RTP-PDU 00.00.41 00.00.47 Send time 00.00.51 Play time

  8. Jitter (contd.) 00.00.00 00.00.10 First RTP-PDU(0) 00.00.11 00.00.20 Second RTP-PDU(10) 00.00.21 00.00.30 00.00.18 00.00.25 00.00.28 Third RTP-PDU(20) 00.00.40 00.00.35 00.00.37 00.00.38 Fourth RTP-PDU (30) 00.00.41 00.00.47 Send time 00.00.48 00.00.51 Play time 00.00.58

  9. Jitter (contd.) Playback buffer At time 00:00:18 At time 00:00:28 At time 00:00:38

  10. Seq no.1, Tmpst 100 Seq no.2, Tmpst 200 Seq no.3, Tmpst 300 sender receiver silence Seq no.4, Tmpst 600 Seq no.5, Tmpst 700 How does Sequence number and Timestamp help ? Audio silenceexample: • Consider audio data • What should the sender do during silence? Not send anything • Why might this cause problems? • Receiver cannot distinguish between loss and silence Solution: • After receiving no PDUs for a while, next PDU received at the receiver will reflect a big jump in timestamp, but have the correct next seq. no. Thus, receiver knows what happened.

  11. Streaming performance requirements • Sequencing – to report PDU loss • to report PDU reordering • to perform out-of-order decoding • Time stamping and Buffering • for play out • for jitterand delay calculation • Payload type identification • for media interpretation • Error concealment –covers up errors from lost PDU by using redundancy in most-adjacent-frame • Quality of Service (QoS) feedback – from receiver to sender for operation adjustment • Rate control –sender reduces sending rate adaptively to network congestion

  12. Support from transport layers • TCP is not used because: TCP does retransmissions  unbounded delays No provision for time stamping TCP does not support multicast TCP congestion control (slow-start) unsuitable for real-time transport RTP + UDP usually used for multimedia services

  13. Protocol stack for multimedia services RTSP RTP RTCP TCP (till now)

  14. RTP: Introduction • Provides end-to-end transport functions for real-time applications • Supports different payload types • All RTP and RTCP PDUs are sent to same multicast group (by all participants) • All RTP PDUs sent to an even-numbered UDP port, 2p • All RTCP PDUs sent to UDP port 2p+1 • Does NOT provide timely delivery or other QoS guarantees • Relies on other protocols like RTCP and lower layers • Does NOT assume the underlying network is reliable and delivers PDUs in sequence • Uses sequence number Application RTP RTCP Transport layer UDP IP Data Link Physical

  15. RTP Session • RTP sessionis sending and receiving of RTP data by a group of participants • Foreach participant, a session is a pair of transport addresses used to communicate with the group • If multiple media types are communicated by the group, the transmission of each medium constitutes a session.

  16. RTP Synchronization Source • synchronization source - each source of RTP PDUs • Identified by a unique,randomly chosen 32-bit ID (theSSRC) • A host generating multiple streams within a single RTP must use a different SSRC per stream

  17. RTP Basics of Data Transmission RTP PDUs

  18. RTP PDU Header Sampling instant of first data octet • multiple PDUs can have same timestamp • not necessarily monotonic • used to synchronize different media streams Incremented by one for each RTP PDU: • PDU loss detection • Restore PDU sequence Payload type Identifies synchronization source Identifies contributing sources (used by mixers)

  19. Mixer RTP mixer - an intermediate system that receives & combines RTP PDUs of one or more RTP sessions into a new RTP PDU • Stream may be transcoded, special effects may be performed. • A mixer will typically have to define synchronization relationships between streams.Thus… • Sources that are mixed together become contributing sources (CSRC) • Mixer itself appears as a new source having a new SSRC

  20. end system 1 from ES1: SSRC=6 from ES2: SSRC=23 from ES1: SSRC=6 transl.1 transl.2 authorized tunnel from ES2: SSRC=23 end system 2 from ES1: SSRC=6 from ES2: SSRC=23 firewall Translator • An intermediate system that… • Connects two or more networks • Multicasting through a firewall • Modifies stream encoding, changing the stream’s timing • Transparent to participants • SSRC’s remain intact

  21. RTP Control Protocol (RTCP) • RTCP specifies report PDUs exchanged between sources and destinations of multimedia information • receiver reception report • sender report • source description report • Reports contain statistics such as the number of RTP-PDUs sent, number of RTP-PDUslost, inter-arrival jitter • Used by application to modify sender transmission rates and for diagnostics purposes

  22. RTCP message types Typically, several RTCP PDUs of different types are transmitted in a single UDP PDU

  23. NTP Timestamp, most significant word NTP Timestamp, least significant word RTP Timestamp Sender’s PDU Count Sender’s Octet Count Sender/Receiver report PDUs V P RC Length (16 bits) PT=200/201  SR/RR Header SSRC of Sender Sender Info SSRC_1 (SSRC of the 1st Source) Fraction Lost Cumulative Number of PDU Lost Extended Highest sequence Number Received Report Block 1 Interarrival Jitter Last SR (LSR) Delay Since Last SR (DLSR) Report Block 2 SSRC_2 (SSRC of the 2nd Source) …… Profile-Specific Extensions

  24. Basic header Ethereal capture for RTP-PDU

  25. header of SR report sender info receiverreport block SDES items Ethereal capture for RTCP-PDU

  26. Synchronization of streams using RTCP RTP audio RTCP audio RTP video RTP video Internetwork • Timestamps in RTP PDUs are tied to the individual video and audio sampling clocks • timestamps are not tied to the wall-clock time, or each other! • Each RTCP sender-report PDU contains (for most recently generated PDU in associated RTP stream): • The timestamp of RTP PDU • The wall-clock time for when PDU was created • Receivers can use this association to synchronize the playout of audio and video

  27. RTCP bandwidth scaling Problem • What happens when there is one sender and many receivers? • RTCP reports scale linearly with the number of participants and would match or exceed the amount of RTP data! More overhead than useful data! Example • Suppose one sender, sending video at a rate of 2 Mbps. Then RTCP attempts to limit its traffic to 100 Kbps. • The 75 kbps is equally shared among receivers: • With R receivers, each receiver gets to send RTCP traffic at 75/R kbps. • Sender gets to send RTCP traffic at 25 kbps. Solution • RTCP attempts to limit its traffic to 5% of the session bandwidth to ensure it can scale! • RTCP gives 75% of this rate to the receivers; and the remaining 25% to the sender.

  28. Real-Time Streaming Protocol (RTSP) • Application layer protocol (default port 554) • Usually runs on RTP for stream & TCP for control • Provides the control channel • Uses out-of-band signaling • Usable for Live broadcasts / multicast Also known as “Network remote control” for multi-media servers.

  29. Web Server web browser HTTP presentation descriptor Presentation descriptor Web Server/Media server RTSP media player pres. desc,streaming commands RTP/RTCP audio/video content RTSP Overview

  30. RTSP Methods

  31. RTSP server TCP get UDP port data source UDP RTCP media server media player RTSP Session Default port 554 RTSP SETUP RTSP OK RTSP PLAY RTSP client RTSP OK RTSP TEARDOWN RTSP OK choose UDP port RTP VIDEO AV subsystem RTP AUDIO

  32. Media server A audio.example.com Media server V video.example.com Client C Web server W -holds the media descriptors Example:Media on demand (Unicast)

  33. C -> W : GET/Twister.sdp HTTP/1.1 Host: www.example.com Accept: application/sdp W-> C : HTTP/1.0 200 OK Content-Type: application/sdp W V C-> A : SETUP rtsp://audio.example.com/twister/audio.en RTSP/1.0 Cseq:1 Transport : RTP/AVP/UDP;unicast;client_port=3056-3057 A-> C : RTSP/1.0 200 OK Cseq:1 Session: 12345678 Transport : RTP/AVP/UDP;unicast;client_port=3056-3057 server_port=5000-5001 C A C->V: SETUP rtsp://video.example.com/twister/video.en RTSP/1.0 Cseq:1 Transport : RTP/AVP/UDP;unicast;client_port=3058-3059 A-> C : RTSP/1.0 200 OK Cseq:1 Session: 23456789 Transport : RTP/AVP/UDP;unicast;client_port=3058-3059 server_port=5002-5003 RTSP Message sequence

  34. C->V: PLAY rtsp://video.example.com/twister/video RTSP/1.0 Cseq: 2 Session: 23456789 V->C: RTSP/1.0 200 OK Cseq: 2 Session: 23456789 RTP-Info: url=rtsp://video.example.com/twister/video; seq=12312232; W V C A C->A: PLAY rtsp://audio.example.com/twister/audio.en RTSP/1.0 Cseq: 2 Session: 12345678 A->C: RTSP/1.0 200 OK Cseq: 2 Session: 12345678 RTP-Info: url=rtsp://audio.example.com/twister/audio.en; seq=876655; RTSP Message sequence (contd.)

  35. C->A: TEARDOWN rtsp://audio.example.com/twister/audio.en RTSP/1.0 Cseq: 3 Session: 12345678 A->C: RTSP/1.0 200 OK Cseq: 3 W V C->V: TEARDOWN rtsp://video.example.com/twister/video RTSP/1.0 Cseq: 3 Session: 23456789 V->C: RTSP/1.0 200 OK Cseq: 3 C A RTSP Message sequence (contd.)

  36. References [1] B. A. Forouzan, “TCP/IP Protocol Suite”, Third edition, [2] H. Schulzrinne, S. Casner, R. Frederick and V. Jacobson, "RTP: a transport protocol for real-time applications", RFC 3550, July 2003. [3] H. Schulzrinne, A. Rao and R. Lanphier, "Real Time Streaming Protocol (RTSP)", RFC 2326, April 1998.

  37. RTCP PDU 1 RTCP PDU 2 sender report receiver report receiver report SR SDES SSRC SSRC SSRC SSRC CNAME PHONE source 2 source 3 compound PDU (single UDP datagram) RTCP compound PDU

  38. RTCP PDU sender report receiver report receiver report SR SSRC SSRC SSRC source 2 source 3 Example source 1 reports, there are 2 other sources

  39. RTCP processing in Translators • SR sender information : Does not generate their own sender information(most of the times), but forwards the SR PDUs received from one side to other • RR reception report blocks : Does not generate their own RR reports (most of the times), but forwards RR reports received from one side to another. SSRC are left intact • SDES : Forwards without changing the SDES info. but may filter non CNAME SDES, if bandwidth is limited • BYE : Forwards BYE PDU unchanged. A translator about to cease forwarding, send a BYE PDU to each connected nodes

  40. RTCP processing in Mixers • SR sender information : Generates its own SR info. Because the characteristics of source stream is lost in the mix. The SR info is sent in same direction as the mixed stream • RR reception report blocks : Generates its own reports for sources in each cloud and sends them only to same cloud • SDES : Forwards without changing the SDES info. but may filter non CNAME SDES, if bandwidth is limited • BYE : Forwards BYE PDU unchanged. A mixer about to cease forwarding, send a BYE PDU to each connected nodes

  41. CNAME=1 length user and domain name Source description PDUs May contain: • a CNAME item (canonical identifier/name) • a NAME item (real user name) • an EMAIL item • a PHONE item • a LOC item (geographic location) • a TOOL item (application name) • a NOTE item (transient msg, e.g. for status) • a PRIV item (private extension) Value 1 2 3 4 5 6 7 8

More Related