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Agenda

A brief overview to RTP and RTCP

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Real-Time Transport Protocol (RTP)

Protocol to the continuous transmission of audiovisual data (streams) over IP-based networks

Product of Internet Engineering Task Force, AVT WG

Standardized first in 1996 RFC 1889

In 2003, a revised RFC RFC 3550 which obsoletes RFC 1889.

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RTP goals

lightweight: specification and implementation

flexible: provide mechanism, don’t dictate algorithms

protocol-neutral: UDP/IP, ST-II, IPX, ATM-AALx, . . .

scalable: unicast, multicast from 2 to O(10^7)

separate control/data: some functions may be taken over by conference control protocol

secure: support for encryption, possibly authentication

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RTP overview

RTP is a packet-based protocol usually running over UDP

Transport multimedia data streams (audio, video, text, etc.) over networks Encode Package Send

It applies in many areas IP telephony technologies (H.323, SIP)

RTP is mainly used for transmission of realtime sensitive data flows, while the Real-Time Streaming Protocol (RTSP) manages and controls the data transmission.

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Architecture

:

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The RTP header

Ver. (2 bits) Indicates the version of the protocol. Current version is 2.

P  (1 bit) Indicates if there are extra padding bytes at the packet.

X  (1 bit) Indicates if protocol extensions are being used in the packet.

CC  (4 bits) Number of CSRC identifiers that follow the fixed header.

M  (1 bit) If it is set, data has some special relevance for the application.

PT  (7 bits) Format of the payload and determines its interpretation by the application.

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What else

Synchronisation Source  The data source, characterized by an identifier (32 bit) in the header

Mixer Switching center, which receive RTP packets from one or multiple sources and

forward them. Mix packages (and synchronizes) and can carry out format conversions.

Translator Don‘t modify packets, but route through firewalls etc.

Reciever The reciever of the RTP packets sorts them on the basis of the sequence

numbers and submits them to the respective application.

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Real Time Control Protocol (RTCP)

Dient der Aushandlung und Einhaltung von Quality of Service (QoS) Parametern durch den periodischen Austausch von Steuernachrichten zwischen Sender und Empfänger

Mittels:

1.Rückmeldung der bisher erbrachten Dienstqualität, wodurch eine Anpassung der Übertragungsrate erfolgen kann

2.Identifikation aller Sitzungsteilnehmer, wodurch semantisch zusammenhängende aber getrennt gesendete Medienströme synchronisiert werden können

3.Steuerung der für RTCP-Pakete verwendeten Bandbreite, damit der Austausch von RTCP-Nachrichten nicht die Übertragung behindert, was bei vielen Sitzungsteilnehmern passieren könnte.

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RTCP packet structure

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RTCP sender reports (SR)

SSRC of sender: identifies source of data

NTP timestamp: when report was sent

RTP timestamp: corresponding “RTP time” lip sync

sender’s packet count: total number sent

sender’s octet count: total number sent

followed by zero or more receiver report

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RTCP receiver reports (RR)

SSRC of source: identifies who’s being reported on

fraction lost: binary fraction

cumulative number of packets lost: long-term loss

highest sequence number received: compare losses, disconnect

interarrival jitter: smoothed interpacket distortion

LSR: time last SR heard

DLSR: delay since last SR

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Questions

Find more on the online slides

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Agenda

Streaming Motivation

The RTSP/RTP Session Overview

Bits from example Session

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RTSP

Wikipedia:

The Real Time Streaming Protocol (RTSP), developed by the IETF and created in 1998 as RFC 2326, is a protocol for use in streaming media systems which allows a client to remotely control a streaming media server, issuing VCR-like commands such as "play" and "pause", and allowing time-based access to files on a server.

The sending of streaming data itself is not part of the RTSP protocol. Most RTSP servers use the standards-based RTP as the transport protocol for the actual audio/video data, acting somewhat as a metadata channel.

…

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Resources on the Web

Wikipedia is a good start:

http://en.wikipedia.org/wiki/Real_Time_Streaming_Protocol

RFC RTSP: http://tools.ietf.org/html/rfc2326

RFC RTP: http://tools.ietf.org/html/rfc3550

Tutorial on RTSP:

http://www.myiptv.org/Articles/RTSP/tabid/72/Default.aspx

Pictures and parts of text taken from:

http://www.cs.columbia.edu/~hgs/rtsp/

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Streaming vs. Television

- reuse existing infrastructure

- flexible media: modem, wireless, cable, LAN, . . .

- Quality scales from stamp-size flipbook to HDTV – adaptive

- Side information easy (closed captioning)

- Easy integration with WWW

- Security through encryption

- Cheap authoring, service lots of content

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Applications

- Lectures, seminars

- On-demand instruction

- Entertainment: specialty content (cf. youtube)- Voice mail

- P2P video streaming

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Challenges

bandwidth: 64–128 kb/s for talking heads, 1.5 Mb/s for movies

quality: packet loss, predictability

reliability: makes CATV look good. . .

billing infrastructure: pay-per-view?

cheap receivers: shouldn’t cost more than set-top box

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Internet real-time & multimedia protocols

resource reservation: RSVP, …

media transport: RTP, AVP, …

stream control: RTSP

stream description: SDP, …

SDP: http://tools.ietf.org/html/rfc4566

AVP: http://www.ece.osu.edu/~zheng/publications/AVP-protocol.pdf

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RTSP and HTTP: similarities

Protocol format: text, MIME-headers

Request/response = request line + headers + body

Status codes

Security mechanisms

URL format

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RTSP is not HTTP

- Server state needed

- Different methods

- Server client

- Protocol Extension posssible:

Extensions start with “x-”, e.g.: x-historyLog

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Agenda

Streaming Motivation

The RTSP/RTP Session Overview

Bits from example Session

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RTSP / RTP Session

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RTSP URLs

Whole presentation:

rtsp://media.example.com:554/twister

track within presentation:

rtsp://media.example.com:554/twister/audiotrack

but: name hierarchy != media hiearchy != file system

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RTSP Request headers

Accept media description formats

Accept-Encoding encoding of media format

Accept-Language human language

Authorization basic and digest authentication

Bandwidth client bandwidth available

Conference conference identifier

From name of requestor

If-Modified-Since conditional retrieval

Range time range to play

Referer how did we get here?

Scale (play time)/(real time)

Speed speed-up delivery

User-Agent software

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RTSP Response headers

Location redirection

Proxy-Authenticate authenticate to proxy

Public methods supported

Retry-After busy, come back later

Server server software

Vary cache tag

WWW-Authenticate request authorization

Hint:

All headers and bodies end with a double carriage return line feed.

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Time Formats

smpte=0:10:00 “Society of Motion Picture and Television Engineers” - hour:minute:second:frame - optional

npt=126.315 Normal play time - {hh;mm;ss}.ms

- mandatory

Rtptime= 1324433 Part of RTP, needed to differentiate between emission time and media time

seq=1214324432part of RTP != time; sequence number of first frame

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Agenda

Streaming Motivation

The RTSP/RTP Session Overview

Bits from example Session

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Unicast session: open streams

C->S: SETUP rtsp://audio.com/twister/audio.en RTSP/1.0CSeq: 1Transport: RTP/AVP/UDP;unicast;client_port=3056-3057

S->C: RTSP/1.0 200 OKCSeq: 1Session: 12345678Transport: RTP/AVP/UDP;unicast;client_port=3056-3057;server_port=5000-5001

C->S: SETUP rtsp://video.com/twister/video RTSP/1.0CSeq: 1Transport: RTP/AVP/UDP;unicast;client_port=3058-3059

S->C: RTSP/1.0 200 OKCSeq: 1Session: 23456789Transport: RTP/AVP/UDP;unicast;client_port=3058-3059;server_port=5002-5003

Error Codes listed in RFC, but same as HTTP

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Unicast session: play

C->S: PLAY rtsp://video.com/twister/video RTSP/1.0CSeq: 2Session: 23456789Range: smpte=0:10:00-

S->C: RTSP/1.0 200 OKCSeq: 2Session: 23456789Range: smpte=0:10:00-0:20:00RTP-Info: url=rtsp://video.com/twister/video;seq=12312232;rtptime=78712811

C->S: PLAY rtsp://audio.com/twister/audio.en RTSP/1.0CSeq: 2Session: 12345678Range: smpte=0:10:00-

S->C: RTSP/1.0 200 OKCSeq: 2Session: 12345678Range: smpte=0:10:00-0:20:00RTP-Info: url=rtsp://audio.com/twister/audio.en;seq=876655;rtptime=1032181

RTP Interaction

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Summary

- Internet multimedia-on-demand - Integrated services - Protocol Extensions

- Conferencing

- Telephony

- WebTV as VOD

- Flexible plug and play RTSP/RTP/AVP/SDP ….