RTP_RTCP.ppt
- 1. RTP/RTCP – Real Time Transport Protocol/ Real Time Control Protocol
- 2. Motivation • TCP/IP – Not suitable for real-time • Retransmissions can lead to high delay and cause delay jitter – Does not support multicast – Congestion control mechanism (slow start) not suitable for AV media • UDP/IP – No defined technique for synchronizing – Streams from different servers may collide – A feedback channel must be defined for quality control
- 3. RTP/RTCP • A session consists of an RTP/RTCP pair of channels • Usually works over UDP/IP • End-to-end protocol
- 4. RTP/RTCP • RTP Features – Multicasting – Payload type identification – Time shaping – Sequencing – Delivery monitoring
- 5. RTP/RTCP • RTP Issues – No QoS guarantees – No guarantee of packet delivery • RTP Timestamp (TS) and Sequence Number (SN) – TS used to order packets in correct timing order – SN to detect packet loss – For a video frame that spans multiple packets – TS is same but SN is different
- 6. RTP/RTCP • RTCP – Synchronize across different media streams – Provide feedback on the quality of data using lost packet counts – Identify and keep track of participants – Retransmission requests
- 8. RTP/RTCP(RFC 1889) • Real-time transport protocol (RTP) is the de facto standard media transport protocol in the Internet – Media transport: audio, vedio, etc. How is it different from data transport? – RTP: media transport. – RTCP: end-to-end monitoring and data delivery and QoS.
- 9. • RTP and RTCP are commonly built on top on UDP. • RTP provides some functionality for media transport. RTP does not: – Guarantee QoS – address resource reservation – Perform signaling (negotiate the media format) • RTP/RTCP are usually implemented within applications --- not as clean cut as TCP.
- 10. • RTP use scenarios – A working group obtains an IP multicast address and a pair of ports through some allocation mechanism. – One port for audio, one for control – The address and the ports are distributed to intended participants. – The participants send audio data in small chucks, 20ms duration. Each chuck of audio data is preceded with a RTP header, which indicates the type of encoding. – The Internet may occasionally lose or reorder packets or delay the packet in variable amount time. To help the receiver reconstruct the timing produced by the sender, RTP header also include timing and sequence number information. – Since the member may join and leave a group dynamically, it is useful to know who is participating and the audio quality at a given time. This task is done through RTCP.
- 11. • RTP packet format: V=2 p x cc m PT sequence number timestamp synchronization source identifier (SSRC) contributing source identifiers (CSRC) p: Padding (1 bit), data content is less than the size of the packet. x: extension(1 bit), fixed header is followed by 1 variable sized header extension. cc: CSRC count: number of CSRC identifiers. m: marker, identify significant events. PT: payload type (PCM, ADPCM, etc)
- 12. • RTP packet format: V=2 p x cc m PT sequence number timestamp synchronization source identifier (SSRC) contributing source identifiers (CSRC) Timestamp: sampling instant of the first octet in the RTP data packet SSRC: sender CSRC list: contributing sources for the payload contained in the packet. (needed for mixers)
- 13. • RTCP – A companion control protocol to RTP. – RTCP is not a signaling protocol – RTCP is used to collect end-to-end information about the quality of the session to each participant. – RTCP packet types: • SR: sender report • RR: receiver report • SDES: Source DEScription • BYE: Hangs up from a session • APP: Application-Sprcific packet
- 14. • SR: sender report – Carry statistics from the active senders. V=2 p rc PT=SR=200 length SSRC of sender NTP timestamp, MSW NTP timestamp, LSW RTP timestamp Sender’s packet count Sender’s octet count SSRC_1 (SSRC of first source), block 1 Sender’s packet count fraction lost cumulative number of packets lost extended highest sequence number received interarrival jitter last SR (LSR) delay since last SR (DLSR) SSRC_2 (SSRC of second source), begin block 2 …. profile-specific extensions
- 15. • RR: receiver report – Statistics from active receivers V=2 p rc PR=RR=201 length SSRC of sender SSRC of first source, begin block 1 Fraction lost (FL) cumlative number of packets lost extended highest sequence number received interarrival jitter last SR (LSR) delay since last SR (DLSR) SSRC of second source, begin block 2 ……. profile-specific extensions
- 16. • SDES: Source Description packet – Allow the binding of SSRC value with an actual identification of the user. – End points need to send an SDES packet at the beginning of the session • BYE: ends a user’s participation in a call. • APP: Application-specific RTP packet.
- 17. • Some issues in RTP and RTCP: – How often should RTCP reports to exchanges. • Must be able to provide real time and with enough resolution for QoS interpretation. • Fix bandwidth • Proportion to channel bandwidth – How to use the RTCP reports? • Reroute – End-to-end QoS information may not be sufficient. • Application adaptation.