rtp tutorial

RTP: Multimedia Streaming over IP

Colin PerkinsUSC Information Sciences Institute

Copyright © 2002 Colin Perkins

Internet Multimedia

• Internet Multimedia has long history:– RFC 741, "Network Voice Protocol", 1977– First video experiments in the early 1980s

• Modern standards development began in 1992:– Developing from teleconferencing systems– Audiocast of IETF meetings

• 20 sites on 3 continents• Precursors to RTP and the present standards

– Standardized RTP in 1996

• Widespread availability of suitable networks in the last couple of years

• Unusual characteristics, worth exploring


Talk Outline

• Use of IP for real-time traffic• Protocols for real-time multimedia over IP

– Outline of signalling protocols– Outline of media transport protocols

• RTP: Real-time Transport Protocol– RTP data transfer protocol– RTP control protocol

• Robustness– Playout and timing correction– Error correction– Security– Congestion control

• Security• Conclusions


The IP Protocol Stack

IPEthernet

ADSLPPP

Optical FibreTwisted Pair

TCP UDP

HTTPFTP

SMTP RTPSIP RTSP

HTML MIME Media codecs

Application programs

Wireless

• Applications can't see the link layers– Just see the performance of

the IP layer– Must assume lowest common

denominator behaviour

• Link layer can't tell the needs of the application– Just see a series of packets– Optimisations for particular

traffic classes are risky– Is the traffic really what you

think?

• Decoupling applications from the network

• IP forms an abstraction layer– Applications and transport

protocols above– Assorted link technologies

below


IP Service Model

The IP service model is limited• Best effort packet transport• Fragmentation• Routing and addressing


Best Effort Packet Transport

• Performance not guaranteed• Packets can be…

– lost– delayed– reordered– duplicated– corrupted

…and the transport protocol must compensate

• Checksum to catch bit errors

• Many causes of problems:– Congestion may cause loss– Packet corruption may cause loss– Route changeover may cause loss– Queuing delay– Multi-path IP routing may reorder– Link-layer striping may reorder– Spurious retransmission and

router bugs cause duplicates

Assumption: Significant packet loss


Packet Loss Patterns

0

5

10

15

20

25

30

35

40

45

8:00 10:00 12:00 14:00 16:00 18:00

Pack

et L

oss

Rat

e (p

erce

nt)

Time of day


Timing Disruption

Rec

eption t

ime

Transmission time

45º line: clocksare synchronized

Discontinuity dueto route change.

Queuing jitter causesvariation in inter-packetarrival time.

Non-45º slope showspresence of clock skew


Best Effort Transport

• Performance can be bad– Applications should be prepared to compensate– Doesn't have to be!

• Loss and jitter can be made arbitrarily low through careful engineering– Most backbone networks have very good performance– Interconnects and customer LANs are currently the main

trouble spots– Explicit QoS doesn't appear necessary


Fragmentation

• IP fragments packets that exceed the MTU– Often >1500 octets

• This causes an undesirable loss multiplier effect– Loss of one fragment means the others must also be

discarded– Better for the application to fragment, and generate small

IP packets

Reconstructed

Fragments

Packet loss

Original


Routing and Addressing

• The use of IP with best effort transport implies heterogeneity

• Multicast makes this an order of magnitude worse– Each receiver sees different

loss characteristics– Hard to get timely feedback

in a scalable manner

• Most communication is unicast– Point-to-point

• Inefficient for many receivers– The server must generate n

copies of the data, for n clients– Significant scaling bottleneck

• Multicast allows the server to scale– Adds network complexity


Properties of IP: Summary

• Best effort, unreliable, packet delivery service– Much of the network performs well– Peering points and customer premises often lacking

• Provides a fragmentation service– Typically best avoided

• Provides a multicast service– In some parts of the network

• 10% of broadband connections

• At best: simple, effective, service


Transport Protocols

• The IP service, by itself, is very limited– Just (tries to) deliver packets

• Always augmented by a transport protocol– TCP– UDP– (others in development)


TCP

• Enhances the raw IP service– Point-to-point and connection oriented– Service selection through ports– Reliable, in-order, delivery– Rate adaptation to match network capacity


TCP

Time

Conges

tion W

indow

Siz

e

Slow start Slow start Congestionavoidance

Congestionavoidance

• Rate adaptation matches throughput to capacity– High link utilization– Fair share between flows

• Retransmission ensures that no data is lost– Reliable, in-order, delivery


UDP

• Uses the services of IP– best effort (unreliable) but timely– fragmentation– routing and addressing– unicast and multicast

• Provides ports, in addition to IP addressing, but no other services


Reliability/Timeliness Tradeoff

Reliable Unreliable

Not timely Timely

TCP UDP

RTP

• Protocols built on unreliable packet networks must make a fundamental trade-off:– Timely– Reliable

• TCP is at one extreme, UDP the other

• Multimedia systems choose their transport carefully:– TCP for signalling– UDP for media data

• Application level protocols can blur the boundary– E.g. RTP for multimedia data – Essential for performance


Talk Outline




• Robustness– Playout and timing correction– Error correction– Security– Congestion control



The Multimedia Protocol Framework

Two fundamental components of the framework:• Signalling protocols• Media transport protocols


Signalling Protocols

The first stage of any multimedia session is signalling• User location• Session initiation, call setup and teardown• Media negotiation• Conference control

Many signalling protocols exist• Teleconferencing: H.323 • Telephony: SIP• Streaming: RTSP, SAP


Signalling Protocols: H.323

• Original signalling protocol for IP-based multimedia– Extension of H.320 ISDN conferencing to IP– Tightly coupled, small group, video conferencing

• Flexible media negotiation and call control• Reputation for complexity

– ASN.1 encoding– Many RTT call setup

• (mostly fixed in later versions)


Signalling Protocols: SIP

• RFC 2543 (recently updated)

• SIP is used to invite someone to join a session– Media negotiation– User location– Call setup and teardown

• Considerable overlap with H.323– More flexible integration with other Internet services

• Email, Web, streaming media, recording, agents, etc.

– More limited media negotiation and call control• Extensions underway

– Very different style• Protocol operation is based on HTTP• Reuses much existing infrastructure


Signalling Protocols: RTSP

• RFC 2326

• Designed for control of a media on demand server– Point-to-point VCR-style remote control– Record/play/rewind/fast-forward

• Widespread commercial use…– RealAudio, QuickTime

• May also be useful for controlling other devices– Voice mail– Interactive voice response

• Leverages HTTP and SIP infrastructure


Signalling Protocols: SAP

• RFC 2974

• A multicast announce-listen protocol for wide area announcement of multimedia sessions– Announcers periodically multicast SDP descriptions to a

well known group– Inter-announcement interval is 10+ minutes– Listeners slowly build up a cache of sessions

• Suitable for announcing long-lived public sessions– E.g. radio/TV station, event coverage

• Mostly used with IP multicast– Talk of use with cable networks


Media Transport Protocols

Once the session has been setup, media flows

Convergence on a single media transport protocol for:• Voice over IP• Teleconferencing• Streaming media

Real-time Transport Protocol, RTP• Flexible, supports many codecs and media types• Extensible to new scenarios


The Multimedia Protocol Framework

Media

codecs

H225.0

IP

UDP TCP

RTP H.245

RASMedia negotiation

Call control

TCP UDP

SIP SAP RTP

Media

codecsLight weight

sessionsMedia negotiation

Call control

IP

IETF Multimedia Protocol Stack ITU Teleconferencing Protocols

RTSP


Design Choices

• Depending on the scenario, implement:– An appropriate signalling protocol

• RTSP• SIP/H.323• SAP

– Media transport using RTP• One or more codecs

– MPEG– H.263

• Error correction and concealment• Congestion Control


Talk Outline




• Robustness– Playout and timing correction– Error correction– Congestion control



RTP: Real-time Transport Protocol

• The standard for real-time transport over IP networks– Streaming audio and video– Voice over IP

• Published as an IETF proposed standard– RFCs 1889 and 1890 in January 1986– Adopted by ITU as part of H.323– Adopted by 3GPP for next generation cellular telephony– Widespread use in streaming: QuickTime, Real, Microsoft

• (HTTP streaming still common)

• Recently revised for draft standard status– Work complete, awaiting publication– Changes include:

• Clarifications and bug-fixes based on experience• Scalability improvements• Support for new codecs

– 100% backwards compatible


Philosophy of RTP

• The challenge: – build a mechanism for robust, real-time media delivery

above an unreliable and unpredictable transport layer – without changing the transport layer

Push responsibility for media

delivery onto the end-points

where possible

Make the system robust to

network problems; media

data should be loss tolerant

The end-to-end argument Application level framing


The End-to-end Argument

• Two options for ensuring reliability– Pass responsibility hop-by-hop, along with the data

• Email

– Responsibility remains with the end points, which ensure delivery even if the intermediate steps are unreliable

• Both TCP and RTP take the second approach

• Consequences:– Intelligence tends to "bubble-up" the protocol stack to the

end points– The intermediate systems can be simple, and need not be

robust• They can simply discard data they cannot deliver, since it will

be recovered end-to-end

• The network is dumb, but end-points are smart


Application Level Framing

• Only the application has sufficient knowledge of its data to make an informed decision about how that data should be transported

• Implications:– The transport protocol should accept data in application

meaningful chunks ("ADUs")• The application must understand the data,• The application must be able to process ADUs independently, in

arbitrary order, and in the presence of loss

– The transport protocol should expose details of delivery, allowing the applications to react intelligently if there are problems

• Blind retransmission is not always appropriate• Maybe the data is stable, and an updated version can be sent• Maybe the data is obsolete, and doesn't need to be resent• Maybe an alternate representation of the data can be sent


Philosophy of RTP

• The philosophy of RTP implies smart, network-aware, applications that are capable of reacting to problems end-to-end.– Both sender and receiver are intelligent– The network is dumb and can be unreliable

• Similar principles apply to the signalling protocols– Mostly end-to-end operation, limited support for network

state

• Fits well with the IP service• Contrast with traditional applications:

– Telephone network is smart, end-points are dumb– MPEG sender is smart, receiver relatively dumb


Protocol Components

RTP Data Transfer Protocol

RTP ProfilePayload

FormatPayload

FormatPayload

FormatPayload

Format

RTP Control Protocol

UDP

IP


Protocol Components


• Provides:– Source identification– Payload identification– Media sequencing– Timing recovery

• Extensions allow for error correction

• Transports application data units– Audio– Video

• One RTP stream transports each media type


Protocol Components

RTP Control Protocol

• Reception quality feedback– Packet loss fraction– Average timing jitter

• Optional source description– Name, location, email

address, phone number

• Mapping from media clock to external time-base– E.g. for lip synchronization

• Loosely coupled membership management


Protocol Components

Payload

FormatPayload

FormatPayload

FormatPayload

Format

• Provide the adaptation layer between a particular codec and RTP

• Optimised for robustness to packet loss

• Many payload formats exist, with more being developed:– H.261, H.263, M-JPEG,

MPEG-2, MPEG-4, BT.656, SMPTE-292


Protocol Components

RTP Profile

• Define use of RTP in particular application scenarios– "Reasonable defaults"– Adaptation to unusual conditions

• Single source multicast• Operation without back channel• Authenticated and secure operation

• Provides a namespace for payload formats


Combining the Pieces

• A multimedia session comprises several RTP sessions– One for each media type

• Each RTP session:– Implements a particular RTP profile– Includes an RTP data flow

• Transporting a single media type according to one or more payload formats

– E.g. Audio switching between G.729 and Fax– E.g. Video using MPEG

– Includes an RTP control protocol flow• Providing reception quality feedback, user information, etc.

– Is defined by:• Source and destination IP addresses• A pair of UDP ports: one for RTP, one for RTCP



• The RTP Data Transfer Protocol delivers a single media stream from sender to one, or more, receivers– Few assumptions about the

underlying transport– Usually runs over UDP/IP

• Typically implemented in an application or as a library– User level, not part of the

kernel

RTP provides:• Source identification• Media identification • Media transport

– Padding, if necessary– Marking of significant

events

• Sequencing• Timing recovery


Packet Format

Payload data

Padding

Synchronization source (SSRC) identifier

Timestamp

Sequence NumberPTMP XV CC

Contributing source (CSRC) identifiers


Source Identification

• Each packet carries a 32 bit synchronization source– Randomly chosen at start-

up, with collision detection

• Provides a transport layer independent identifier– Supports gateways– IPv4, IPv6, ATM

• Identifies a single time-synchronized media flow– Mapped to a persistent

identifier using RTCP

Payload data

Padding

Timestamp





Source Identification

• Each packet may include a list of contributing sources– Allows data from up to 16

sources to be identified– Each CSRC is the SSRC of a

mixed participant

• Allows RTP to support mixers and translators– Mixers combine several flows

into one• E.g. Conferencing MCU

– Translators change the format of a flow, or gateway between different networks

• Transcode to a lower bit-rate• Gateway between unicast

and multicast

Payload data

Padding

Timestamp

Sequence NumberPTMP XV


CC



Communication Models

• Mixers and translators greatly expand the range of communications models available to RTP

RTP end point

RTP Translator or mixer

Multicast group, thenetwork replicates data as necessary, withno translation or mixing.

Point-to-point communication

via unicast

Four participants communicating via a multicast group

Replicated unicast: agroup of three using anRTP translator/mixer tomediate communications.

Translated: multicast to unicast. Two participantscommunicating via a multicast group, with a third linked to the sessionby an RTP translator.


Media Identification

• Each packet carries a 7 bit payload type field

• Mapped to a payload formatduring session setup– Allows flexible signalling of

codec type and parameters– Mapping can be static, if the

profile allows

• Each flow carries only one type of media– Only audio, or only video

• The payload type allows the sender to switch between a set of payload formats– E.g. a flow carries only audio

but may switch between fax and voice at any time

Payload data

Padding

Timestamp

Sequence NumberMP XV CC



PT


Media Transport and Payload Formats

• Packets contain a block of payload data, described by a payload format

• Payload formats describe the mapping between codec output and RTP packets– Chosen so that each packet is

independently decodable– Application level framing

• The payload data typically includes a payload header to ease parsing– E.g. The H.261 payload format

copies some information from the GOB header so each set of macro-blocks can be decoded independently


Payload data

Padding

Timestamp

Sequence NumberMP XV CC


PT


Media Transport: Marker

• Each packet includes a bit to mark significant events– Start of talk spurt for audio– Last packet of frame for video

• A hint that special processing may be required Payload data

Padding

Timestamp

Sequence NumberPTP XV CC



M


Media Transport: Padding

• Each packet may be padded beyond its natural size

• Rarely used, but needed by some encryption algorithms– DES in CBC modes operates

on 64 bit blocks

• The SRTP profile provides a better security solution

Payload data

Timestamp

Sequence NumberPTMXV CC



P

Padding


Sequencing

• Each packet contains a 16 bit sequence number– Random initial value– Increments monotonically

with each packet sent– Wraps around to zero when

the limit is reached

• Used to detect packet loss– Is not used to determine

playout order

• Basic RTP does not provide error correction– The receiver is expected to

conceal the error, and to continue processing

– Extensions provide forward error correction and limited retransmission

Payload data

Padding

Timestamp

PTMP XV CC



Sequence Number


Timing Recovery

• Each packet contains a 32 bit timestamp

• Indicates the sampling instant of the oldest payload data– Determines playout order

• The clock rate is defined by the payload format:– Audio clock is sampling rate– Video clock is 90kHz,

indicating the frame time– Mapping to codec time-base

is also defined

• No requirements on stability or accuracy of clock– Implies receiver adaptation

• Time code not carried directly, but mapping to wall clock time via RTCP sender reports

Payload data

Padding




Timestamp


RTP Control Protocol (RTCP)

• Each RTP data flow has an associated control flow• The control flow provides:

– Time-base management– Quality of service feedback– Member identification and management


Time-base Management

• Timestamps map between the RTP timeline and NTP “wall-clock” time– If a common NTP clock is used for multiple streams, a

receiver can synchronize them

• No explicit transport of SMPTE (or similar) time-codes – Can be derived from NTP timestamps

• Accuracy limited by NTP resolution, unless external clock provided

• RTSP provides a mapping function

• Also allows receivers to estimate data/packet rate and possibly clock skew


Reception Quality Reporting

• Quality of service feedback from each receiver:– Loss fraction– Cumulative number of packets lost– Highest sequence number received– Inter-arrival jitter– Round-trip time

• Many uses:– Loss rate can be used to select amount of FEC to employ– Jitter gives estimate of playout buffer delay at receiver


Membership Management

• RTCP provides a canonical name, mapping SSRC to a persistent identifier– Used to associate streams

for synchronisation

• RTCP can optionally deliver source description data:– Name– Email address– Phone number– Location– (extend with metadata)

• Provides loosely coupled presence information– Explicit leave message

• Augments the membership management provided by the signalling protocol– Primarily using the explicit

leave indication


RTCP reporting interval

• RTCP is a low-rate reporting protocol– Not intended for uses that require instant feedback– Scalable to very large sessions

• Statistical summary of group conditions

• Packets are sent periodically– The interval between packets is adjusted to limit RTCP to

once per 5 seconds, or 5% of the data rate– Randomized to avoid synchronization


RTP: Summary

• Flexible and extensible media transfer protocol– Supports a range of codecs– Allows detection of network problems– Allows recovery of media timing

• Associated, low rate, reporting of reception quality, time-base, and presence information


Talk Outline







Robustness

• RTP operates over UDP/IP– Best effort delivery– Packets can be lost, delayed, reordered, duplicated, etc.

• Applications are responsible for correct playout– Timing recovery– Error concealment– Congestion control


Timing Recovery

• The network can seriously disrupt media timing

• Receivers must include a jitter compensation buffer to reconstruct the media for playout

Internet

Router

Original media stream

Sender

Router

Constant inter-packet spacing

Network inducestiming jitter into the media stream

Received media stream

Receiver

Variable inter-packet spacing


Playout and Timing Correction

2nd talk spurt1st talk spurt

Transmission

Reception

Playout

Jitter affects inter-packet timing

NetworkTransit

PlayoutBuffer

Network transit delay

Packet discardeddue to late arrivalPlayout buffering delay added

to compensate for jitter.


Playout and Timing Correction

• RTP does not specify a standard playout buffer or timing reconstruction algorithm– Provides the necessary

information; allowing product differentiation

• Compare with MPEG, where the buffer model is closely defined

• Many trade-offs to consider:– latency versus quality– speed of reaction to change– buffering ability

• Typical design:– Streaming applications use

large delay (10+ seconds)– Interactive applications try

to keep delay low (tens of milliseconds)


Error Correction

• Limited Retransmission– RTCP feedback profile

• Forward Error Correction– Media specific– Media independent

• Unequal error protection• Interleaving

RTP

Reliable

Not timely Timely

Unreliable

TCP UDP

⇒ Add limited reliability to RTP


Retransmission in RTP

• Why must retransmission be limited?– Timely versus reliable– We don't want to re-invent TCP

• How to implement retransmission?– RTCP provides a back channel– Modify the RTCP timing rules to allow early feedback

• Keep the fundamental scaling rules to avoid potential for implosion


RTCP Feedback Profile

• RTCP reports sent as usual• Feedback can be sent early

– Ignore 5 second rule– Borrow bandwidth from the

next reporting interval– Delays next report– Send minimal report packet

• Transport layer feedback– NACK, ACK sequence number

• Payload specific feedback– Reference picture selection– MPEG-4 NEWPRED

• Under development in IETF

Groupsize

2

RegularRTCP mode

Early RTCPmode

ImmediateFB mode

Report everyrelevant eventimmediately

Report manyof the events

but not all

Just regularRTCP packets

Send feedback + regular RTCP packets


Forward Error Correction

• Retransmission relies on feedback from receivers– Must request that lost

packets are resent

• Works well in many cases• Two scenarios where it is

inefficient:– Round-trip time is large– Many receivers, independent

loss events

• Alternative:– Sender adds redundant data

to the media stream– Receivers use this to correct

errors, without contacting sender

• Forward Error Correction– Well known technique at the

link layer– Also be applicable at the IP

level and above

RTP RTP RTP FEC RTP RTP RTP FEC RTP


Media Specific FEC

• Some codecs may naturally be loss tolerant

• Design payload format to take advantage of this


Media Specific FEC

• Some codecs may naturally be loss tolerant

• Design payload format to take advantage of this– Audio/video redundancy– RFC 2198, AMR, H.263+

1 2 3 4

1 2 3 4

1 2 4

1 2 3 4

Original Stream

Redundant data added

Packet lost in transport

Reconstructed stream


Media Independent FEC

• Media specific FEC needs to be produced by the encoder– Part of the payload format– Either off-line or on-line

⇒ Compression performed at time of transmission⇒ To pick appropriate FEC⇒ Undesirable⇒ Too much load on sender to

support many streams

• Better if the FEC can be derived from pre-compressed media– Less load on sender– Perhaps less network efficient

Source image

Compress Packetize

RTP RTP

FEC

FECRTP RTP


Parity FEC

• Parity codes to protect serial communication well known

• Can apply same technique to packet networks– Generate parity packet

• Standard for parity FEC:– RFC 2733– Flexible parity operation

• Standards for Reed-Solomon coding under development

1 1 1 0 0 0 1

0 0 1 1 0 1 0

Bit stream A

Bit stream B

1 1 0 1 0 1 1Parity code: A XOR B

1 1 1 0 0 0 1

1 1 0 1 0 1 1

Transmission loses B

0 0 1 1 0 1 0Calculate parity to recover B = A XOR (A XOR B)


Unequal Error Protection

• Not all data in the packets is equally important– Headers and codec state

updates are vital– Media data is of variable

importance• Data used for predication• Data used in a single frame

• Some links have high bit error rate– Causes packet corruption– Detected by UDP checksum– Packet discarded

• Seriously impacts wireless link performance– Cellular, especially

Partial Checksumat the UDP level


Interleaving

• Can interleave, to make bursts of loss appear as random loss– Adds considerable delay

• Popular with streaming apps– Part of many audio payload

formats

• Packet loss concealment and correction work best when loss is isolated– Single packet losses

• Packet loss on the Internet is bursty

1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 16

1 5 9 13 2 6 10 14 3 7 11 15 4 8 12 16

1 2 4 5 6 8 9 10 12 13 14 16

1 5 9 13 2 6 10 14 4 8 12 16

Original stream

Interleaved stream

Packet loss

Reconstructed stream


Error Correction: Summary

⇒ Network doesn't have to be perfect• If it's okay for data traffic,

it's probably okay for video• Unless you have very strict

requirements

⇒ Acceptable to overprovision for QoS• No real need for RSVP or

differentiated services

⇒ Well designed applications can tolerate significant loss• Often, 5% loss acceptable

• Extensive, and ongoing, research and standards work developing error correction for RTP– Retransmission– Media/codec specific FEC– Media independent FEC– Partial checksum– Interleaving


Congestion Control

But… … the preceding assumed traffic is well behaved

… assumed flows respond to congestion in the network

… or, QoS and flow admission control employed


Congestion Control

• An IP network provides a best-effort packet-switched service. – No admission control– The network accepts all

packets and tries to deliver them.

• However, no guarantee of delivery provided– Excess packets discarded if

links become congested.

• The transport protocol must detect loss, and reduce its rate to allow the congestion to clear– TCP does this automatically– RTP does not

Packets sent

Pack

ets d

eliv

ered

Normal operation Congestion Collapse


Congestion Control

• Adaptation must be done by the application– Possible rate changes

depend on the codec– Complex feedback loop

between codec and network

• For RTP, implies senders should observe receiver feedback– If loss fraction is non-zero,

consider sending less– As loss decreases, consider

sending faster

Packets sent

Pack

ets d

eliv

ered

Normal operation Congestion Collapse


TCP Friendly Rate Control

• Possible to predict the long-term average throughput of TCP

• Derive throughput in terms of observed loss rate, RTT and packet size– Measurable qualities in RTP

• Adapt sending rate to match– Driven by reception of RTCP– Drives codec operation

• The "best current practice" congestion control scheme for multimedia flows

• With appropriate parameters:– Fair to TCP on average– Slowly changing rate8

3)321(33

2 2 pTpppR

sT

rto⋅++=


Limitations of Congestion Control

• TCP friendly algorithms are new, and evolving– Very limited deployment – Not clear that they have

reached their final form

• Interactions between codec and network are not well defined– Unclear how slow response,

or limited adaptability, will impact fairness

• Human factors aspects play a key role– Congestion control implies

variable quality– Subjectively very annoying,

unless the rate of change is slow

– Can have a significant impact on congestion control


Talk Outline







Security

Several aspects to multimedia security• Confidentiality of the media• Authentication of the sender• Watermarking• Storage

RTP can help here


Security in RTP

• Basic RTP provides limited security– Packets may be encrypted

• DES is specified; algorithm may be negotiated during session setup

– Does not support sender authentication

• A secure RTP profile is under development– Encryption for confidentiality

• Encrypts only the payload data, not the headers

• AES in counter or F8 mode• Robust to bit errors, allows

header compression, suitable for cellular wireless

– Sender authentication• Adds a trailer to each packet,

containing authentication code• HMAC-SHA1


Watermarking, Storage and DRM

• RTP is concerned only with the transmission of media

• Does not consider:– The contents of the video

image– How the image is captured,

generated and stored

• Implications:– Receivers may store the payload

• RTP cannot influence the process• The copy may not be perfect

– Watermarks or other embedded data can be inserted at source

• But must be robust to packet loss


Talk Outline







Conclusions

• IP provides a non-optimal service for video transport– Careful network engineering can solve many problems

• Multimedia protocol framework comprises:– Signalling: H.323, SIP, RTSP, SAP– Media Transport: RTP + codecs

• RTP provides:– Robust, flexible and extensible media transport– Range of error correction schemes– Range of security solutions

• Limitations: – Congestion control


For More Information

IETF Audio/Video Transport Working Grouphttp://www.ietf.org/html.charters/avt-charter.html

Colin Perkinshttp://www.east.isi.edu/~csp/

http://www.ietf.org/html.charters/avt-charter.html

http://www.ietf.org/html.charters/avt-charter.html

http://www.east.isi.edu/~csp/

rtp tutorial

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