RTP

RTP: Real-timeTransport Protocol

Presented by:1- AbdElrahman Mohamed Amer

2-Tarek Nader Esmat

Presentation Outline

Introduction History of streaming media Overview of RTP What is RTP? Why RTP? How RTP work? RTP Features (Advantages) RTP Disadvantages RTP HeaderReal-time streaming protocol (RTSP)

Brief history of streaming media

Origin of Protocol(RTP)

Developed by the Audio-Video Transport Working Group of the IETF “Internet Engineering Task Force” : H. Schulzrinne (Columbia University) S. Casner (Packet Design) R. Frederick (Blue Coat Systems Inc.) V. Jacobson (Packet Design)

First published as RFC 1889 in the year 1996

Superseded by RFC 3550 in the year 2003

Real-Time Transport Protocol

End-to-end delivery services for applications transmitting real-time data, such as audio and video over multicast and unicast networks.

Applications that use RTP are:Less sensitive to packet loss.Very sensitive to packet delays.

E.x:-– Video teleconferencing– Internet Telephony (VoIP)– Internet audio, video streaming

Application layer protocol

Typically used on top of IP and UDP•Layer 4 – UDP or TCP•Layer 3 – Internet Protocol IP•Layer 7 – RTP

How RTP work?

RTP Features All RTP and RTCP PDUs are sent to same

multicast group (by all participants)

RTP supports several file formats like MPEG and MJPEG but you can add extra formats.

RTP also has jitter compensation, multimedia streaming (tolerates packet loss) and detection of out of sequence arrival in data attributes.

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 jitter and 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

Ideal Timing – no jitter

00.00.00

00.00.10

00.00.20

00.00.30

00.00.11

00.00.21

00.00.31

Send time

Play time

30 s

econ

ds

First RTP-PDU

Second RTP-PDU

Third RTP-PDU

application

Reality – jitter

00.00.00

00.00.10

00.00.20

00.00.30

00.00.11

Send time

Play time

00.00.21

00.00.25

00.00.35

00.00.37

00.00.47

delay

First RTP-PDU

Second RTP-PDU

Third RTP-PDU 00.00.40

Fourth RTP-PDU 00.00.41

00.00.51

Jitter (contd.)

00.00.00

00.00.10

00.00.20

00.00.30

00.00.11

Send time

Play time

00.00.21

00.00.25

00.00.3500.00.37

00.00.47

First RTP-PDU(0)

Second RTP-PDU(10)

Third RTP-PDU(20) 00.00.40

Fourth RTP-PDU (30) 00.00.41

00.00.51

00.00.18

00.00.28

00.00.38

00.00.48

00.00.58

Jitter (contd.)

Playback buffer

At time 00:00:18

At time 00:00:28

At time 00:00:38

Support from transport layersWhy not only RTP?

Assume the underlying network is reliable and delivers PDUs in sequence. Uses sequence number.

Provide timely delivery or other QoS guarantees. However, it does provide necessary data to application to order packets and adjust signal quality. Relies on other protocols like RTCP and lower layers (e.g. UDP, TCP) to do so.

Handle resource reservation .

Need other protocols like RSVP

Handle connection setups or tear-downs.

Need other protocols like SIP or H.323

Define media data formats or encodings.

Need media specific profiles.

RTP Does Not

RTCP

Why not UDP?

UDP offers datagram-like serviceConnectionless , Unreliable , Unordered No Flow , Error , Congestion controlPort numbers

RTP + UDP usually used for multimedia services

Why not TCP?

TCP does retransmissions unbounded delays due to (Acks, Flow control, windowing) No provision for time stamping TCP does not support multicast TCP congestion control (slow-start) unsuitable for real-time transport (A-V media)

RTP Packets

RTP PDU HeaderIncremented by one for each RTP PDU:

PDU loss detection

Restore PDU sequence

Sampling instant of first data octet

multiple PDUs can have same timestamp

not necessarily monotonic

used to synchronize different

media streams

Payload type

Identifies synchronization source

(used by mixers)Identifies contributing sources

V - Version. 2 bits. X - Extension. 1 bit. CC - CSRC count. 4 bits.

M - Marker. 1 bit. e.g., the start of a talk spurt

PT - Payload Type. 7 bits. e.g., specific audio codec• PT=0: PCM 64Kbpz; PT=3: GSM 13Kbps

Sequence Number. 16 bits. (packet count)

Timestamp. 32 bits.

SSRC, Synchronization source. 32 bits. CSRC, Contributing source. 32 bits.

P- padding .1bit

- gap in sequence #: packet loss

• e.g., audio@8KHz, 20ms samples/packet- timestamp increment per packet: 160

• 90KHz used for video- a video frame may be encapsulated in a few packets

• gap in timestamp: silence

RTP Data Transfer Protocol : Synchronization

RTP Data Transfer Protocol : RTP profiles

Media specific (e.g., audio)

• timestamp: e.g., sampling rate, 8KHz PCM• packet size: e.g., about 20ms samples in PCM

- packets independent as much as possible: ALF

SSRC, Synchronization source. 32 bits.

Sequence Number. 16 bits.

Timestamp. 32 bits.

M - Marker. 1 bit. PT - Payload Type. 7 bits.

Ethereal capture for RTP-PDU

Basic header

RTP Session

RTP session is sending and receiving of RTP data by a group of participants

For each 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.

RTP Synchronization Source

synchronization source - each source of RTP PDUs

Identified by a unique,randomly chosen 32-bit ID (the SSRC)

A host generating multiple streams within a single RTP must use a different SSRC per stream

RTP Basics of Data Transmission

RTP PDUs

How does Sequence number and Timestamp help ?

Audio silence example:

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.

– Why might this cause problems?

send

er

rece

iver

silence

Seq no.1, Tmpst 100Seq no.2, Tmpst 200

Seq no.3, Tmpst 300

Seq no.4, Tmpst 600Seq no.5, Tmpst 700

Consider audio data– What should the sender do during silence?

Not send anything

Receiver cannot distinguish between loss and silence

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.

web browser

media player

Web Server

Web Server/Media server

RTSP Overview

RTSPpres. desc,streaming commands

RTP/RTCPaudio/video content

Presentation

descriptor

HTTPpresentation descriptor

RTSP Methods

OPTIONSC S

determine capabilities of server/clientC S

DESCRIBE C S get description of media stream

ANNOUNCE C S announce new session description

SETUP C S create media session

RECORD C S start media recording

PLAY C S start media delivery

PAUSE C S pause media delivery

REDIRECT C S redirection to another server

TEARDOWN C S immediate teardown

SET_PARAMETER C S change server/client parameter

GET_PARAMETER C S read server/client parameter

RTSP Session

media server

RTSPserver

datasource

media player

AVsubsyste

m

RTSPclient

RTSP OK

RTSP PLAY

RTSP OK

RTP AUDIO

RTP VIDEO

RTSP TEARDOWNRTSP OK

get UDP portchooseUDP port

RTSP SETUP

Default port 554

RTCP

TCP

UDP

Example:Media on demand (Unicast)

Media server A

audio.example.com

Media server V

video.example.com

Web server W

-holds the media descriptors

Client C

RTSP Message sequence

C

W

V

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


server_port=5002-5003

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

C-> A : SETUP rtsp://audio.example.com/twister/audio.en RTSP/1.0

Cseq:1


A-> C : RTSP/1.0 200 OK

Cseq:1

Session: 12345678


server_port=5000-5001

RTSP Message sequence (contd.)

C

W

V

A

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;

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.)

C

W

V

A

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

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

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.

QUESTIONS?

Thank you for your attention!

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