RTP

RTP/RTCP/RTSP

Real-Time Protocols

Presented by-1. A. H. M. Kamal (0411054002)2. KAZY NOOR –E- ALAM SIDDIQUEE (0411052009)3. MASUD4.Mahmudul Hasan Razib (0411052060)

Overview

• Streaming performance requirements• Protocol stack for multimedia services• Real-time transport protocol (RTP)• RTP control protocol (RTCP)• Real-time streaming protocol (RTSP)

Real-time multimedia streaming• Real-time multimedia applications

– Video teleconferencing– Internet Telephony (VoIP)– Internet audio, video streaming

(A-PDUs)

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

eco

nds

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 bufferAt time 00:00:18

At time 00:00:28

At time 00:00:38

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? sen

der

rece

iver

silence

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

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

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

Support from transport layers

Not like streaming stored audio and videosupport longer delay?

• Video conference• Internet Telephony

Real time interactive audio video

TCP(till now)

RTSP

Protocol stack for multimedia services

RTP RTCP

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

RTP RTCP

Application

UDP

IP

Data Link

Physical

Transport

layer

RTP Session

For an RTP session

Single multicast address

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

RTP Packet

RTP Header Audio/Video Chunk

RTP Packet

RTP Header Audio/Video Chunk

RTP Header

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

RTP Payload-Type

Payload-Type Number Audio Format Sampling Rate (kHz) Rate (kbps)

0 PCM u-low 8 64

1 1016 8 4.8

3 GSM 8 13

7 LPC 8 2.4

9 G.722 16 48-64

14 MPEG Audio 90 -

15 G.728 8 16

Audio Payload Type

Video Payload Type

Payload Type Number Video Format

26 Motion JPEG

31 H.261

32 MPEG 1 video

33 MPEG 2 video

MixerRTP 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

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

end system 1

end system 2

transl.1from ES1: SSRC=6

from ES2: SSRC=23transl.2

from ES2: SSRC=23from ES1: SSRC=6

authorized tunnel

firewallfrom ES2: SSRC=23from ES1: SSRC=6

RTP: Developing Software App. with RTP

RTP RTCP

UDP

IP

Data Link

Physical

Transport

layer

Application

RTP

Socket

UDP

IP

Data Link

Physical

• Do by hand

• Add RTP header to frame

• Pass to UDP socket

Application

• Use JAVA Class/ C RTP Library

• Send a chunk of media, Payload type no., SSRC, timestamp to interface

Approach One Alternative Approach

RTP Security

Confidentiality

Authentication and Integrity

Allows encription to RTP Packets

Believes to lower levels to ensure authentication.

Integrity by sanity checking descriptor header: protocol version no, pkt length, payload type.

Key Management Conference type App:

Combination of SIP, SAP, SDP

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-PDUs lost, inter-arrival jitter

Feedback Information: modify sender transmission rates and for diagnostics purposes-whether problems are local, regional, global

• 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!

Synchronization of streams using RTCP

• 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

Internetwork

RTP audio

RTCP audio

RTP video

RTP video

RTCP bandwidth scaling

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.

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.

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!

RTCP bandwidth scaling

Sender Can Send T=# of Senders

.25x.05xSession BWAvg. RTP packet size

Receiver Can Send T=# of Senders

.75x.05xSession BWAvg. RTP packet size

Real-Time Streaming Protocol (RTSP)

• Application layer protocol (default port 554)• Usually runs on RTP for stream & TCP for control• Provides the control channel to transmit• 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

•Reference to several continuous media file

•Directive for synchronization

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

RTCP compound PDU

SRsenderreport

receiverreport

receiverreportSS

RC

SS

RC

SS

RC

source 2 source 3

RTCP PDU 1

SDES CNAME PHONE

SS

RC

RTCP PDU 2

compound PDU(single UDP datagram)

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

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

Source description PDUsMay 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

CNAME=1 length user and domain name

Here is the end of my presentation on RTP

=Thanks to all=

Documents

RTP