RFC 1889

Network Working Group
Request for Comments: 1889
Category: Standards Track

Audio-Video Transport Working Group
H. Schulzrinne
GMD Fokus
S. Casner
Precept Software, Inc.
R. Frederick
Xerox Palo Alto Research Center
V. Jacobson
Lawrence Berkeley National Laboratory
January 1996

Status of this Memo

This document specifies an Internet standards track protocol for the Internet community, and requests discussion and suggestions for improvements. Please refer to the current edition of the "Internet Official Protocol Standards" (STD 1) for the standardization state and status of this protocol. Distribution of this memo is unlimited.

Abstract

This memorandum describes RTP, the real-time transport protocol. RTP provides end-to-end network transport functions suitable for applications transmitting real-time data, such as audio, video or simulation data, over multicast or unicast network services. RTP does not address resource reservation and does not guarantee quality-of- service for real-time services. The data transport is augmented by a control protocol (RTCP) to allow monitoring of the data delivery in a manner scalable to large multicast networks, and to provide minimal control and identification functionality. RTP and RTCP are designed to be independent of the underlying transport and network layers. The protocol supports the use of RTP-level translators and mixers.

Table of Contents

• 1. Introduction
• 2. RTP Use Scenarios
  • 2.1 Simple Multicast Audio Conference
  • 2.2 Audio and Video Conference
  • 2.3 Mixers and Translators
• 3. Definitions
• 4. Byte Order, Alignment, and Time Format
• 5. RTP Data Transfer Protocol
  • 5.1 RTP Fixed Header Fields
  • 5.2 Multiplexing RTP Sessions
  • 5.3 Profile-Specific Modifications to the RTP Header
    • 5.3.1 RTP Header Extension
• 6. RTP Control Protocol -- RTCP
  • 6.1 RTCP Packet Format
  • 6.2 RTCP Transmission Interval
    • 6.2.1 Maintaining the number of session members
    • 6.2.2 Allocation of source description bandwidth
  • 6.3 Sender and Receiver Reports
    • 6.3.1 SR: Sender report RTCP packet
    • 6.3.2 RR: Receiver report RTCP packet
    • 6.3.3 Extending the sender and receiver reports
    • 6.3.4 Analyzing sender and receiver reports
  • 6.4 SDES: Source description RTCP packet
    • 6.4.1 CNAME: Canonical end-point identifier SDES item
    • 6.4.2 NAME: User name SDES item
    • 6.4.3 EMAIL: Electronic mail address SDES item
    • 6.4.4 PHONE: Phone number SDES item
    • 6.4.5 LOC: Geographic user location SDES item
    • 6.4.6 TOOL: Application or tool name SDES item
    • 6.4.7 NOTE: Notice/status SDES item
    • 6.4.8 PRIV: Private extensions SDES item
  • 6.5 BYE: Goodbye RTCP packet
  • 6.6 APP: Application-defined RTCP packet
• 7. RTP Translators and Mixers
  • 7.1 General Description
  • 7.2 RTCP Processing in Translators
  • 7.3 RTCP Processing in Mixers
  • 7.4 Cascaded Mixers
• 8. SSRC Identifier Allocation and Use
  • 8.1 Probability of Collision
  • 8.2 Collision Resolution and Loop Detection
• 9. Security
  • 9.1 Confidentiality
  • 9.2 Authentication and Message Integrity
• 10. RTP over Network and Transport Protocols
• 11. Summary of Protocol Constants
  • 11.1 RTCP packet types
  • 11.2 SDES types
• 12. RTP Profiles and Payload Format Specifications
• A. Algorithms
  • A.1 RTP Data Header Validity Checks
  • A.2 RTCP Header Validity Checks
  • A.3 Determining the Number of RTP Packets Expected and Lost
  • A.4 Generating SDES RTCP Packets
  • A.5 Parsing RTCP SDES Packets
  • A.6 Generating a Random 32-bit Identifier
  • A.7 Computing the RTCP Transmission Interval
  • A.8 Estimating the Interarrival Jitter
• B. Security Considerations
• C. Authors' Addresses
• D. Bibliography

• Original text document
• Complete HTML RFC (TAR, TGZ, or ZIP format)

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