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Nguyen Thi Mai Trang Lip6/Phare Thi-Mai-Trang - Nguyen@

RTP is a standard protocol for transporting real-time data such as audio and video over IP networks. It addresses issues with using TCP or UDP alone for real-time media. RTP works with RTCP for quality of service reporting and synchronization of multiple media streams. RTP uses UDP with sequence numbers and timestamps to allow reordering of packets and synchronization of different media streams. RTCP periodically reports reception quality and participant identification to support an RTP session.

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Ahmed TeeJay
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54 views

Nguyen Thi Mai Trang Lip6/Phare Thi-Mai-Trang - Nguyen@

RTP is a standard protocol for transporting real-time data such as audio and video over IP networks. It addresses issues with using TCP or UDP alone for real-time media. RTP works with RTCP for quality of service reporting and synchronization of multiple media streams. RTP uses UDP with sequence numbers and timestamps to allow reordering of packets and synchronization of different media streams. RTCP periodically reports reception quality and participant identification to support an RTP session.

Uploaded by

Ahmed TeeJay
Copyright
© Attribution Non-Commercial (BY-NC)
Available Formats
Download as PDF, TXT or read online on Scribd
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RTP

Nguyen Thi Mai Trang LIP6/PHARE [email protected]

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Plan
Introduction RTP RTCP

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Introduction
Problems with telephony over the Internet
Quality of service is not guaranteed (delay, bandwidth, data loss, jitter)

Why another transport protocol for real-time data?


TCP can introduce high delay because of retransmission UDP does not allow the application to reorder voice packets because of the lack of sequence number
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RTP (1)
Real-time Transport Protocol, RFC 3550 Standardized in 2003 by the IETF (Internet Engineering Task Force) Protocol for real-time data transport over the Internet
Voice-over-IP Telephony Teleconferencing Streaming video
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RTP (2)
Client Server

CODEC RTP UDP IP Physical IP Physical

CODEC RTP UDP IP Physical

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RTP packet generation


Application (Audio/video CODECS)

RTP

RTP

RTP

RTP UDP

RTP UDP
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RTP UDP
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RTP sessions (1)


A session consists of a group of participants who are communicating using RTP A participant may be active in multiple RTP sessions (e.g. one for audio and one for video) For each participant, the session is identified by a network address, a port pair for sending data and a port pair for receiving data
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RTP sessions (2)


Each port pair comprises two adjacent ports
An even-numbered port for RTP data packets The next higher (odd-numbered) port for RTCP control packets

A session can be unicast or multicast RTP translator or mixer can be used in a session to adapt the data transmission to participants conditions

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RTP sessions (3)


RTP end point Translator/mixer Multicast group Multicast Unicast

Unicast with mixer/translator Translated, multicast to unicast


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RTP packet format


V P X CC M PT Timestamp Synchronization source (SSRC) identifier Contributing source (CSRC) identifiers (if mixers are used) Header extension (optional) Payload header (payload format dependent) Sequence number V = version P = Padding X = Extension CC = CSRC count M = Marker PT = Payload Type

Payload data Padding

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RTP header fields (1)


Version (2 bits)
RTP version (version 2 currently)

Padding (1 bit)
1: The packet contains at least one octet of padding at the end and the last octet of the RTP packet contains the size of the padding part

Extension (1 bit)
1: one experimental header extension is appended at the end of the header

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RTP header fields (2)


CSRC count (4 bits)
Contains the number of CSRC identifiers that follow the fixed header

Marker (1 bit)
The interpretation of the Marker bit is defined by a profile

Payload type (7 bits)


Identifies the media transported by an RTP packet

Sequence number (16 bits)


Increased by one for each RTP packet sent Used by the receiver to detect packet loss and restore packet sequence

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RTP header fields (3)


Timestamp (32 bits)
Reflect the sampling instant of the first byte in the RTP data packet Receiver uses timestamp to remove packet jitter and provide synchronous playout

SSRC (32 bits)


Identify the source of the RTP stream Each stream in an RTP session has a distinct SSRC which is a number that the source assigns randomly when the new stream is started

CSRC (32 bits/CSRC, up to 15 CSRCs)


Present only when inserted by a mixer using the SSRC identifiers of contributing sources allowing correct talker indication at the receiver

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Examples of static payload type assignments


Payload Type N 0 3 8 31 32 Payload format AUDIO/PCMU AUDIO/GSM AUDIO/PCMA VIDEO/H261 VIDEO/MPV Specification RFC 1890 RFC 1890 RFC 1890 RFC 2032 RFC 2250 Description ITU G.711 -law audio GSM full rate audio ITU G.711 A-law audio ITU H.261 video MPEG I/II video

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Translator and mixer


Translator
Operate on RTP data while maintaining the synchronization source and timeline of the stream (e.g. converting between media encoding formats without mixing)

Mixer
Receive RTP packets from a group of sources and combine them into a single output, possibly changing the encoding, before forwarding the result

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RTCP (1)
RTP Control Protocol, RFC 3550 Designed to be used with RTP and to control an RTP session RTP and RTCP packets belonging to the same session use the same multicast address but different port number
RTCP port number = RTP port number + 1
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RTCP (2)
Sender
RTP RTCP

Internet Internet
RT CP
P RT

CP RT

RT P

Receiver

Receiver

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RTCP (3)
RTCP packets are sent periodically to provide
Periodic reporting of reception quality (e.g. number of packets sent, number of packets lost, inter-arrival jitter) Participant identification Other source description information Notification on changes in session membership Information needed to synchronize media streams

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RTCP bandwidth scaling (1)


If the number of receiver is large, RTCP has a potential scaling problem RTCP modifies the rate at which a participant sends RTCP packets into the multicast tree as a function of the number of participants in the session

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RTCP bandwidth scaling (2)


RTCP limits its traffic to 5% of the session bandwidth 75% of RTCP traffic is given to receivers and 25% of RTCP traffic is given to the sender The period for transmitting RTCP packets for a sender
T= Number _ of _ senders average _ RTCP _ packet _ size 0.25 0.05 session _ bandwidth

The period for transmitting RTCP packets for a receiver


T= Number _ of _ receivers average _ RTCP _ packet _ size 0.75 0.05 session _ bandwidth

References
C. Perkins, RTP Audio and video for the Internet, Addison-Wesley 2003 J. F. Kurose, K. W. Ross, Computer networking: A top-down approach featuring the Internet, 3rd edition, Addison Wesley 2005

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