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RealReal--Time Transport andTime Transport and

Control ProtocolControl Protocol(RTP/RTCP)(RTP/RTCP)

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OverviewOverview

Introduction to RTP and RTCPIntroduction to RTP and RTCP RealReal--time Transport Protocol (RTP)time Transport Protocol (RTP)

RTP ImplementationsRTP Implementations

RTP in a Multimedia ContextRTP in a Multimedia Context

RTP Profiles and PayloadsRTP Profiles and Payloads

RTP Header FormatRTP Header Format

RTP Header Extension FormatRTP Header Extension Format

RealReal--time Transport Control Protocol (RTCP)time Transport Control Protocol (RTCP)

More on RTCPMore on RTCP

ReferencesReferences

ConclusionsConclusions

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Introduction to RTP/RTCPIntroduction to RTP/RTCP

The Real-time Transport Protocol (RTP) works

alongside with the Real-time Transport Control

Protocol (RTCP). In fact, RTP as a protocol is made up of 2 parts - a

data part (RTP) and a control part (RTCP).

Together, these protocols deliver real-time trafficwith timing information and feedback on the

reception quality.

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RealReal--time Transport Protocol (RTP)time Transport Protocol (RTP)

RTP is a transport protocol that meets the

requirements of voice and other real-time data.

RTP can transport data continuously rather than inbursts, and it can handle data delivery in multicast

environments.

RTP takes care of data that TCP cannot handle.

Such data can include voice, video, etc.

RTP was designed to run independently of the

underlying transport and network layers of the OSI

model.

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Functionalities of RTPFunctionalities of RTP

Timing recoveryTiming recovery

SynchronizationSynchronization

Content identificationContent identification

DemultiplexingDemultiplexing

Loss detection (errorLoss detection (error--recognition)recognition)

Security servicesSecurity services

Applicatio

/

UD

I

Li k layer

hysical layer

ra sport layer{Socket

i terface

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RTP PayloadRTP Payload

Stream TranslationStream Translation

TranslatorTranslator

MixerMixer

P eader RTP ayloadRTP eaderU P eader

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RTP ImplementationsRTP Implementations

Most implementations of RTP run in IP

environments on top of the ser Datagram Protocol

( DP).

But RTP can also be used with other protocols such

as the Asynchronous Transfer Mode (ATM), etc.

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RTP in a Multimedia contextRTP in a Multimedia context

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RTP in a Multimedia ContextRTP in a Multimedia Context

The RTCP packets report on the reception qualityfor a RTP session.

The routers on the network communicate through

RSVP (R

esource ReS

erV

ationP

rotocol) to set asidebandwidth for the transmission of the multimedia

data.

The RTP header provides timing information so

that data delivery can be properly synchronized.

The RTP header also specifies the payload type. In

this way multiple data and data compression types

are allowed.

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RTP Profile and PayloadRTP Profile and Payload

RTP is tailored to the application sending data by

means of supplementary profiles and payload

formats.

A payload format defines how a particular payload

has to be carried in RTP.

A profile assigns payload type numbers for the set

of payload formats contained in the application.Application-specific extensions or modifications to

RTP can also be defined by a profile.

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Required Bandwidth of Some EncodingRequired Bandwidth of Some Encoding

Formats (Throughput)Formats (Throughput)

PT Name of

Encoding Met od

T rougput

[kbit/s]

0 PCM 64

1 1016 4,8

2 G.726-32 32

3 GSM 13

6 DVI4 64

7 LPC 2,4

9 G.722 48-64

15 G.728 16

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The first 12 octets are present in every RTP packet, while the

list of CSRC identifiers is present only when inserted by a mixer.

The fields have the following meaning:

The version (V) field identifies the version of RTP, and it consistsof 2 bits.

The padding (P) field consists of 1 bit. If this bit is set, the packet

contains one or more additional octets at the end which are not part

of the payload. The last octet of the padding contains a count of howmany padding octets should be ignored.

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RTP Header Format (Cont.)RTP Header Format (Cont.)

Padding may be needed by some encryption algorithms with fixedblock sizes or for carrying several RTP packets in a lower-layer

protocol data unit.

The extension (X) field consists of 1 bit and if this bit is set, the

fixed header is followed by exactly one header extension, with a

format defined later.

The CSRC count (CC) field consists of 4 bits and it contains the

number of CSRC identifiers that follow the fixed header.

The marker (M) field consists of 1 bit. The interpretation of the

marker is defined by a profile. It is intended to allow significantevents such as frame boundaries to be marked in the packet stream.

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A profile may define additional marker bits or specify that there isno marker bit by changing the number of bits in the payload type

field.

The payload type (PT) field is of 7 bits and it identifies the RTP

payload format, and determines its interpretation by the application.

The sequence number field consists of 16 bits and it increments

by one for each RTP data packet sent.

The timestamp field which consists of 32 bits reflects the

sampling instant of the first octet in the RTP data packet. The

sampling instant must be derived from a clock that incrementsmonotonically and linearly in time to allow synchronization and

jitter calculations.

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The SSRC field identifies the synchronization source.This identifier is chosen randomly, with the intent that no

two synchronization sources within the same RTP session

will have the same SSRC identifier.

CSRC list: 0 to 15 items, 32 bits eachThe CSRC list identifies the contributing sources for the

payload contained in this packet. The number of

identifiers is given by the CC field. If there are more than

15 contributing sources, only 15 may be identified.

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tkzsi valsznsgChoosing Synchronization Source (SSRC)Choosing Synchronization Source (SSRC)

IdentifiersIdentifiersNumber of sources, L is the length of the identifier (32 bit)

P

Ns

s

L!

1

2

2

1exp

Ns

Its probability that two sources independently get the same SSRC

(in case of large number of sources):

In case of joining new sourc (when the others have already unique

identifier), the collision probability:

Pc $ v2 10 7Ps $

10 4

P Nc sL! 2

For Ns ! 1000 :

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RTP Header Extension FormatRTP Header Extension Format

If the X bit in the RTP header is one, a variable-length

header extension is appended to the RTP header,

following the CSRC list if present. The header extensioncontains a 16-bit length field that counts the number of

32-bit words in the extension, excluding the four-octet

extension header (therefore zero is a valid length).

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RTP Header Extension Format (Cont.)RTP Header Extension Format (Cont.)

Only a single extension may be appended to the RTP

data header. To allow multiple inter-operating

implementations to each experiment independently with

different header extensions, or to allow a particularimplementation to experiment with more than one type

of header extension, the first 16 bits of the header

extension are left open for distinguishing identifiers or

parameters. The format of these 16 bits is to be definedby the profile specification under which the

implementations are operating.

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Application of MPEGApplication of MPEG--H.261 TranslatorH.261 Translator

Lege

Translator

Host

64 k s WAN

Local ampus

Network

G-H.261

ra slator

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RealReal--time Transport Control Protocoltime Transport Control Protocol

As mentioned earlier, RTCP provides information

on the quality of reception of the data transmitted.

This information is very useful to the applications asthey can change the way they behave on basis of

RTCP reports

The RTCP feedback information can also be auseful tool for network managers

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Traffic of RTP and RTCP PackagesTraffic of RTP and RTCP Packages

Sender Receiver

Receiver

Internet

RTCPRTCP

RTCP

RTP

RTP

RTP

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More on RTCPMore on RTCP

RTCP is also used to keep track of the

participants in a RTP session. It does this by

carrying the RTP source identifier, called thecanonical name (CNAME), and the synchronization

source identifier (SSRC).

RTCP also controls the rate at which participants

in a RTP session transmit RTCP packets. The aim ofthis is to keep control traffic to a maximum of 5

percent of overall session traffic.

A RTCP BYE packet is transmitted for

termination.

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Controlling RTCP Transmission Intervals,Controlling RTCP Transmission Intervals,

Bandwidth FittingBandwidth Fitting The control traffic is limited to the maximum 5 % of the wholeThe control traffic is limited to the maximum 5 % of the wholetraffic by the modification of the RTCP package sending ratetraffic by the modification of the RTCP package sending rate

The available bandwidth of the RTCP is allocated to senders andThe available bandwidth of the RTCP is allocated to senders and

receivers, the part of the senders:receivers, the part of the senders:

all receivers together:all receivers together:

At sender: sending interval of RTCP packages:At sender: sending interval of RTCP packages:

number of senders:number of senders:

bandwidth of the whole session:bandwidth of the whole session: BB

average RTCP package size:average RTCP package size: At receiver the sending intervalAt receiver the sending interval

RTCP packages:RTCP packages:

number of receivers:number of receivers:

T pN

B rs

s

s

! 0 05,

pN

B (1r )r

r

s

! 0 05,

rs

! 25%

1 rs

Ns

p

Nr

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Estimation of Packet LossEstimation of Packet Loss

The known expression for the TCP traffic can be used for theestimating the lpacket loss ration of other traffic which uses

the same path

Let c=1,22, RTT=200 ms, the encoding method is DVI (ADPCMversion), 40 ms data packet, RTP/ DP/IP header 40 bytes,

codec state 4 bytes, media data 160 bytes, thus p=204 bytes,

sendign rate 25 packet/s,

thus the bandwith required by the stream:

From these the packet loss ratio: l= 6,0 %

lp c

B RTT!

1

2

Bpacket

s

byte

packet

byte

s1 25 204 5100! v !

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ConclusionsConclusions

RTP and RTCP work together

Together, these protocols deliver real-time

traffic with timing information and feedback on

the reception quality.

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