05. May 20041INF-3190: Multimedia Protocols Multimedia Protocols Foreleser: Carsten Griwodz Email: [email protected].

05. May 2004 1INF-3190: Multimedia

Protocols

Multimedia Protocols

Foreleser: Carsten GriwodzEmail: [email protected]


Protocols

Non-QoSMultimedia Networking

RTP – Real-Time Transfer Protocol


Protocols

Real-time Transport Protocol (RTP)

Real-time Transport Protocol (RTP) RFC 1889 Designed for requirements of real-time data transport NOT real-time NOT a transport protocol

Two Components Real-Time Transfer Protocol (RTP) RTP Control Protocol (RTCP)

Provides end-to-end transport functions Scalable in multicast scenarios Media independent Mixer and translator support RTCP for QoS feedback and session information


Protocols

Real-time Transport Protocol (RTP)

No premise on underlying resources

layered above transport protocol

no reservation / guarantees

Integrated with applications

RTP follows principles of Application Level Framing

and Integrated Layer

Processing

UDP

IPv4/6

EthernetAAL5ATM

ST-2

RTP RTCP

mediaencapsulation

application

TCP

ATM


Protocols

RTP RTP services are

Sequencing Synchronization Payload identification QoS feedback and session information

RTP supports Multicast in a scalable way Generic real-time media and changing codecs on the fly Mixers and translators to adapt to bandwidth limitations Encryption

RTP is not designed for Reliable delivery QoS provision or reservation


Protocols

RTP Functions RTP with RTCP provides

Support for transmission of real-time data Over multicast or unicast network services

Functional basis for this Loss detection – sequence numbering Determination of media encoding Synchronization – timing Framing - “guidelines” in payload format definitions Encryption Unicast and multicast support Support for stream “translation” and “mixing” (SSRC;

CSRC)


Protocols

RTP Packet FormatTypical IETF RFC bit-exact representation

0 1 2 3 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |V=2|P|X| CC |M| PT | SEQ | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | TST | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | synchronization source (SSRC) identifier | +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+ | contributing source (CSRC) identifiers | | .... |

+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+| header extension |

+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+| payload (audio, video, ...) || .... |+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

a byte

a longword (32 bit)


Protocols

RTP Packet Format




4 bit CSRC count, indicates the number of contributing sources in the header

Marker bitMeaning depends on payload

profile,e.g. frame boundary

Version number,Always 2

Padding indicator bitif set, number of padding bytes is in last byte of payload

Header extension bitTrue if header extension is present

7 bit payload typeAllows identification of thepayload’s content type


Protocols

RTP Packet Format




16 bit sequence number

Several 32 bit CSRCContribution source identifier, the number is

indicated by CCA mixer copies the original sources’ SSRCs

here

32 bit timestamp

32 bit SSRCSynchronization source identifier, a random

number identifying the sender

Header extensionmultiples of 32 bit


Protocols

RTP Architecture Concepts Integrated Layer Processing

Typical layered Data units sequentially processed by each layer

Integrated layer processing Adjacent layers tightly coupled

Therefore, RTP is not complete by itself: requires application-layer functionality/information in header




16 bit sequence number32 bit timestamp

7 bit payload typeAllows identification of thepayload’s content type

Marker bitMeaning depends on payload profile, e.g. frame boundary


Protocols

RTP Packet Format Relatively long header (>40 bytes)

overhead carrying possibly small payload header compression other means to reduce bandwidth (e.g. silence

suppression)

No length field Exactly one RTP packet carried in UDP packet Can use TCP or ATM AAL5

do-it-yourself packaging

Header extensions for payload specific fields possible

Specific codecs Error recovery mechanisms


Protocols

RTP Profile (RFC 1890) Set of standard encodings and payload types

Audio: e.g. PCM-u, GSM, G.721 Video: e.g. JPEG, H.261

Number of samples or frames in RTP packet Sample-based audio: no limit on number of samples Frame-based audio: several frames in RTP packet

allowed

Clock rate for timestamp Packetized audio: default packetization interval 20 ms Video: normally 90 kHz, other rates possible


Protocols

RTP Profiles Payload type identification

RTP provides services needed for generic A/V transport Particular codecs with additional requirements Payload formats defined for each codec: syntax and semantic of

RTP payload Payload types

Static: RTP AV profile document Dynamic: agreement on per-session basis

Profiles and Payload Formats in RTP Framework

RTP / RTCP

AV Profile

AdditionalProfiles

PayloadFormats

Dynamic Payload Types

PT mapping outside RTP

(e.g. SDP)


Protocols

RTP Profiles General

Associated with a media type. Provides association between PT field and specific media

format Defines sampling rate of timestamp May also define or recommend a definition for the

“marker” bit

Video Profile Marker bit recommended to mean last packet

associated with a timestamp Timestamp clock: 90000 Hz Defines PT mapping for a number of different video

encoding types


Protocols

RTP Profiles Audio Profile

Marker bit set on the first packet after a silence period where no packets sent

Timestamp equals sampling rate Recommends 20ms minimum frame time Recommends that samples from multiple channels be

sent together Defines PT for a number of different audio encoding

types


Protocols

RTP Profile for MPEG Video Payload

Frame headersGOP header


Protocols


Fragmentation rules Video sequence header

if present, starts at the beginning of an RTP packet GOP sequence header

Either at beginning of RTP packet Or following video sequence header

Picture header Either at beginning of RTP packet Following GOP header

No header can span packets

Marker Bit Set to 1 if packet is end of picture


Protocols


MPEG-1 Video specific payload header

MBZ Must be zero

TR Temporal reference The same number for all

packets of one frame For ordering inside an MPEG

GOP S

1 is sequence header is in this packet

B 1 if payload starts with new

slice E

1 if last byte of payload is end of slice

P 3 bits that indicate picture

type (I,P, or B) FBV, BFC, FFC, FFC

Indicate how a P or B frame is related to other I and P frames

MPEG Video Profile 0 1 2 3 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | MBZ | TR |MBZ|S|B|E| P | | BFC | | FFC | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ FBV FFV


Protocols

RTP Quality Adaptation

Component interoperations for control of quality Evaluation of sender and receiver reports Modification of encoding schemes and parameters Adaptation of transmission rates Hook for possible retransmissions (outside RTP)

Application

RTCP RTP

DecodingEncoding

Application

UDP/IP UDP/IP

RTCPRTP

Decoding Encoding


Protocols

RTP Control Protocol (RTCP) Companion protocol to RTP (tight integration with RTP)

Monitoring of QoS of application performance

Feedback to members of a group about delivery quality, loss, etc.

Sources may adjust data rate Receivers can determine if QoS problems are local or network-wide

Loose session control Convey information about participants Convey information about session relationships

Automatic adjustment to overhead report frequency based on participant count

Typically, “RTP does ...” means “RTP with RTCP does ...”


Protocols

RTCP Packets

Several RTCP packets carried in one compound packet RTCP Packet Structure

SR Sender Report (statistics from active senders:bytes sent -> estimate rate)

RR Receiver Report (statistics from receivers) SDES Source Descriptions (sources as “chunks” with

several items like canonical names, email, location,...)

BYE explicit leave APP extensions, application specific

R SR / RR BYESDES APP

Compound (UDP) Packet


Protocols

RTCP Sender / Receiver Reports Sender report

Sender Information Timestamps Packet Count, Byte Count

List of statistics per source Receiver report

For each source Loss statistics Inter-arrival jitter Timestamp of last SR Delay between reception of last SR

and sending of RR Analysis of reports

Cumulative counts for short and long time measurements

NTP timestamp for encoding- and profile independent monitoring

HeaderSender Information

Reception Report

Profile Specific Extensions

Reception Report ...

Header

Reception Report

Profile Specific Extensions

Reception Report ...


Protocols

RTP Mixer Mixer

Reconstructs constant spacing generated by sender Translates audio encoding to a lower-bandwidth Mixes reconstructed audio streams into a single stream Resynchronizes incoming audio packets

New synchronization source value (SSRC) stored in packet Incoming SSRCs are copied into the contributing

synchronization source list (CSRC) Forwards the mixed packet stream Useful in conference bridges


Protocols

RTP Translator

Translation between protocols e.g., between IP and ST-2 Two types of translators are installed

Translation between encoding of data e.g. for reduction of bandwidth without adapting sources

No resynchronization in translators SSRC and CSRC remain unchanged

ATM UDP

ProtocolTranslator

MPEGSource

MPEGSink

H.263Sink

ProfileTranslator


Protocols

RTP Identifiers

S1 S3

S2 S4

M1 M2T1 R1

S1:10

S2:1

M1:33 (10,1)

M1:33 (10,1)

S4:13

S4:13

S3:19

M2:17 (19,13,33)

SSRC chosen by sender S1

Translators keep SSRCs and CSRCs

SSRC chosen by mixer M1

CSRCs from mixed sources S1 and S2

CSRCs contain previous SSRCs, but not previous CSRCs


Protocols

Protocol Development

Changes and extensions to RTP Scalability to very large multicast groups Congestion Control Algorithms to calculate RTCP packet rate Several profile and payload formats Efficient packetization of Audio / Video RTCP-based retransmission Loss / error recovery


Protocols

Non-QoSMultimedia Networking

Signalling Protocols:RTSP and SIP


Protocols

Signaling Protocols Applications differ

Media delivery controlled by sender or receiver Sender and receiver “meet” before media delivery

Signaling should reflect different needs Media-on-demand

Receiver controlled delivery of content Explicit session setup

Internet telephony and conferences: Bi-directional data flow, live sources (mostly) explicit session setup, mostly persons at both

ends Internet broadcast

Sender announces multicast stream No explicit session setup


Protocols

Real-Time Streaming Protocol (RTSP)

Internet media-on-demand Select and playback streaming media from server Similar to VCR, but

Potentially new functionality Integration with Web Security Varying quality

Need for control protocol Start, stop, pause, …

RTSP is also usable for Near video-on-demand (multicast) Live broadcasts (multicast, restricted control

functionality) ...


Protocols

RTSP Approach In line with established Internet protocols

Similar to HTTP 1.1 in style Uses URLs for addressing:

rtsp://video.server.com:8765/videos/themovie.mpg Range definitions Proxy usage Expiration dates for RTSP DESCRIBE responses Other referenced protocols from Internet (RTP, SDP)

Functional differences from HTTP Data transfer is separate from RTSP connection

typically via RTP Server maintains state – setup and teardown messages Server as well as clients can send requests


Protocols

RTSP Features Rough synchronization

Media description in DESCRIBE response Timing description in SETUP response Fine-grained through RTP sender reports

Aggregate and separate control of streams possible

Virtual presentations Server controls timing for aggregate sessions RTSP Server may control several data (RTP) servers

Load balancing through redirect at connect time Use REDIRECT at connect time

Caching Only RTSP caching so far Data stream caching is under discussion


Protocols

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


Protocols

media server

RTSP Integration

HTTPserver

RTSPserver

datasource

web browser

AVsubsystem

RTSPplug-in

HTTP GETpresentation description file

RTSP SETUPRTSP OK

RTSP PLAYRTSP OK

RTP AUDIO

RTP VIDEO

RTSP TEARDOWNRTSP OK


Protocols

Session Initiation Protocol (SIP) Lightweight generic signaling protocol Internet telephony and conferencing

Call: association between number of participants

Signaling association as signaling state at endpoints (no network resources)

Several “services” needed Name translation User location Feature negotiation Call control Changing features


Protocols

SIP Basics Call user Re-negotiate call parameters Forwarding (manual and automatic) Call center Supports personal mobility (change of terminal)

Through proxies or redirection Terminate / transfer calls ASCII (readable) protocol – SIP vs. H.323

Similarities (request/response, proxies ...) Differences (server state, server may initiate actions ...)

Control, location and media description (via SDP) Extensible towards

Usage for IP-IP, POTS-IP, inter-gateway interaction with firewalls, billing system, ...

Different modes Proxy mode Redirect mode


Protocols

SIP Operation – Proxy Mode

Proxy forwards requests Possibly in parallel to several hosts Cannot accept or reject call Useful to hide location of callee

User with“symbolic name”

calls another

1. Invite u@domain

7. Ok

Site A

8. ACK u@domain

11. Ok

4. Invite user@host

6. Ok

9. ACK user@host

10. Ok

5. “Ring”

2.

Where

?

3.

use

r@h

ost

Proxy ModeSite B

Location Server


Protocols

SIP Operation – Redirect Mode

User with“symbolic name”

calls another

2.

Where

?

3.

use

r@h

ost

Location Server

Redirect Mode1. Invite u@domain

4. Moved temporarilyLocation: user@domain2

5. ACK u@domain

6. Invite user@domain2 7. “Ring”

7. Ok

8. ACK user@domain2

Site A

Site B


Protocols

SIP – Methods Basic Methods (RFC 2543)

TABLE Additional Methods (partially

standardized) INFO: carry information between User Agents REFER: ask someone to send an INVITE to

another participant SUBSCRIBE: request to be notified of specific

event NOTIFY: notification of specific event

05. May 20041INF-3190: Multimedia Protocols Multimedia Protocols Foreleser: Carsten Griwodz Email: [email protected].

Documents

multimedia protocolsforeleser

applications rtp

random number

number of contributing

number of padding bytes

payload profile

frame boundaryversion

multicast supportsupport