Real-time Transport Protocol (RTP)

Emil Diaz VoIP Security COMS 4995 03 Dept. of Computer Science Henning Schulzrinne Columbia University Fall 2008

Overview
Protocol

goals & background Transport Protocol (RTP)

Real-time RTP Use

Control Protocol (RTCP)

Scenarios

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Real-time Transport Protocol

Real-Time Transport Protocol

End-to-end delivery services for applications transmitting real-time data, such as audio and video

Payload type identification Sequence numbering Time stamping Delivery monitoring lightweight flexible scalable
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Goals

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Origin of Protocol

Developed by the Audio-Video Transport Working Group of the IETF:

H. Schulzrinne (Columbia University) S. Casner (Packet Design) R. Frederick (Blue Coat Systems Inc.) V. Jacobson (Packet Design)

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First published as RFC 1889 in the year 1996 Superseded by RFC 3550 in the year 2003
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Protocol Structure

Application layer protocol

Typically used on top of IP and UDP

Applications that use RTP are:

Less sensitive to packet loss Very sensitive to packet delays

UDP provides key services:

Multiplexing Checksum
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RTP Does NOT

Define media data formats or encodings

Need media specific profiles

Handle connection setups or tear-downs

Need other protocols like SIP or H.323

Handle resource reservation

Need other protocols like RSVP

Guarantee timely delivery or Quality of Service

However, it does provide necessary data to application to order packets and adjust signal quality
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RTP Sample Packet

http://www.codeproject.com/KB/IP/Video_Voice_Conferencing.aspx
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RTCP Sample Packet

5 main types:
Sender Report (SR) Receiver Report (RR) Source Description (SDES) Bye Message (BYE)

Application-defined (APP)

http://developer.apple.com/documentation/QuickTime/QTSS/Concepts/chapter_2_section_13.html
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Use Scenarios

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Real-time Transport Protocol

Simple Multicast Audio Conference

1. 2. 3. 4. 5. 6.

7.

Call connection established Audio sampled at 20ms durations Each data chunk is packaged with an RTP header RTP packet is wrapped around UDP packet Sent through network Receiver receives and parses RTP header Based on payload type, application reconstructs audio stream in 20ms chunks
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Audio and Video Conference

Audio and Video has separate RTP/RTCP sessions

Receiver can choose media type to use Encodings can change independently Network resource allocations per media basis

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Mixers and Translators

Accommodate participant network resources

Mixer Low Bandwidth Mixer Combining media streams Translator Forward RTP packets to private networks

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Quality & Data Distribution Feedback

Packet loss, congestion, jitter, delivery times

Directly useful for control of adaptive encodings Identify if problems are local or global Short-term and long-term statistical analysis Each participant eventually knows about the other members Source description dynamically identifies who is sending Active senders get more bandwidth Session bandwidth kept constant by adjusting transmission rate based on the number of participants
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Self-adjusting network

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Source Description Information

CNAME Conical name (joe@example.com) NAME Display Name (Joe Smith) EMAIL Email (different than CNAME) PHONE International phone number LOC Location information (City, Building, Room?) TOOL Application (VideoView 2.0)

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References

RFC 3550 - http://tools.ietf.org/html/rfc1889 RFC 3551 - http://tools.ietf.org/html/rfc3551 RTP News - http://www.cs.columbia.edu/~hgs/rtp/ Wikepedia:

RTP - http://en.wikipedia.org/wiki/Real-time_Transport_Protocol RTCP - http://en.wikipedia.org/wiki/RTCP

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Summary

Services:

payload type identification sequence numbering time stamping delivery monitoring mixers & translators

Goals :

lightweight flexible protocol-neutral scalable separate control/data secure

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