Agenda

1
Agenda

• A brief overview to RTP and RTCP

2
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.

3
RTP goals

• lightweight specification and implementation
• flexible provide mechanism, dont dictate
  algorithms
• protocol-neutral UDP/IP, ST-II, IPX, ATM-AALx, .
  . .
• scalable unicast, multicast from 2 to O(107)
• separate control/data some functions may be
  taken over by conference control protocol
• secure support for encryption, possibly
  authentication

4
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.

5
Architecture
6
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.

7
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
• Dont 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.

8
(No Transcript)
9
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
• Rückmeldung der bisher erbrachten Dienstqualität,
  wodurch eine Anpassung der Übertragungsrate
  erfolgen kann
• Identifikation aller Sitzungsteilnehmer, wodurch
  semantisch zusammenhängende aber getrennt
  gesendete Medienströme synchronisiert werden
  können
• Steuerung der für RTCP-Pakete verwendeten
  Bandbreite, damit der Austausch von
  RTCP-Nachrichten nicht die Übertragung behindert,
  was bei vielen Sitzungsteilnehmern passieren
  könnte.

10
RTCP packet structure
11
RTCP sender reports (SR)

• SSRC of sender identifies source of data
• NTP timestamp when report was sent
• RTP timestamp corresponding RTP time ? lip
  sync
• senders packet count total number sent
• senders octet count total number sent
• followed by zero or more receiver report

12
RTCP receiver reports (RR)

• SSRC of source identifies whos 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

13
Questions

• Find more on the online slides

14
Agenda

• Streaming Motivation

The RTSP/RTP Session Overview
Bits from example Session
15
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.

16
Resources on the Web

• Wikipedia is a good start
• http//en.wikipedia.org/wiki/Real_Time_Streaming_P
  rotocol
• 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/Defau
  lt.aspx
• Pictures and parts of text taken from
• http//www.cs.columbia.edu/hgs/rtsp/

17
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

18
Applications

• - Lectures, seminars
• - On-demand instruction
• - Entertainment specialty content (cf. youtube)
• Voice mail
• - P2P video streaming

19
Challenges

• bandwidth 64128 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 shouldnt cost more than set-top
  box

20
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-pro
tocol.pdf
21
RTSP and HTTP similarities

• Protocol format text, MIME-headers
• Request/response request line headers body
• Status codes
• Security mechanisms
• URL format

22
RTSP is not HTTP

• - Server state needed
• - Different methods
• - Server ? client
• - Protocol Extension posssible
• Extensions start with x-, e.g. x-historyLog

23
Agenda

• Streaming Motivation

The RTSP/RTP Session Overview
Bits from example Session
24
RTSP / RTP Session
25
RTSP URLs

• Whole presentation
• rtsp//media.example.com554/twister
• track within presentation
• rtsp//media.example.com554/twister/audiotrack
• but name hierarchy ! media hiearchy ! file
  system

26
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

27
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.

28
Time Formats

• smpte01000 Society of Motion Picture and
  Television Engineers - hourminutesecondfra
  me - optional
• npt126.315 Normal play time - hhmmss.ms
• - mandatory
• Rtptime 1324433 Part of RTP, needed to
  differentiate between emission time and media
  time
• seq1214324432 part of RTP ! time sequence
  number of first frame

29
Agenda

• Streaming Motivation

The RTSP/RTP Session Overview
Bits from example Session
30
Unicast session open streams

• C-gtS SETUP rtsp//audio.com/twister/audio.en
  RTSP/1.0
• CSeq 1
• Transport RTP/AVP/UDPunicastclient_port3056-
  3057
• S-gtC RTSP/1.0 200 OK
• CSeq 1
• Session 12345678
• Transport RTP/AVP/UDPunicast
• client_port3056-3057server_port5000-5001
• C-gtS SETUP rtsp//video.com/twister/video
  RTSP/1.0
• CSeq 1
• Transport RTP/AVP/UDPunicastclient_port3058-
  3059
• S-gtC RTSP/1.0 200 OK
• CSeq 1
• Session 23456789
• Transport RTP/AVP/UDPunicast
• client_port3058-3059server_port5002-5003

Error Codes listed in RFC, but same as HTTP
31
Unicast session play

• C-gtS PLAY rtsp//video.com/twister/video
  RTSP/1.0
• CSeq 2
• Session 23456789
• Range smpte01000-
• S-gtC RTSP/1.0 200 OK
• CSeq 2
• Session 23456789
• Range smpte01000-02000
• RTP-Info urlrtsp//video.com/twister/videoseq
  12312232rtptime78712811
• C-gtS PLAY rtsp//audio.com/twister/audio.en
  RTSP/1.0
• CSeq 2
• Session 12345678
• Range smpte01000-
• S-gtC RTSP/1.0 200 OK
• CSeq 2
• Session 12345678

RTP Interaction
32
Summary

• - Internet multimedia-on-demand
• Integrated services
• Protocol Extensions
• - Conferencing
• - Telephony
• - WebTV as VOD
• - Flexible plug and play RTSP/RTP/AVP/SDP .