rfc1889
draft-ietf-avt-rtp-new
rfc1890
draft-ietf-avt-profile-new
rfc2198
rfc2833
rfc2793
draft-ietf-avt-ilbc-codec
draft-herlein-speex-rtp-profile
draft-duric-rtp-ilbc
rfc1889
draft-ietf-avt-rtp-new
rfc1890
draft-ietf-avt-profile-new
rfc2198
rfc2833
rfc2793
draft-ietf-avt-ilbc-codec
draft-herlein-speex-rtp-profile
draft-duric-rtp-ilbc
rfc1889.txt  | RTP: A Transport Protocol for Real-Time Applications |
| Author(s) | Audio-Video Transport Working Group, H. Schulzrinne, S. Casner, R. Frederick, V. Jacobson |
| Organization | ietf |
| State | proposed standard |
| Size | 188544 bytes |
| obsoleted by | rfc3550.txt |
| 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. |
| draft-ietf-avt-rtp-new-12.txt
|
"RTP: A Transport Protocol for Real-Time Applications", Van Jacobson,
Stephen Casner, Ron Frederick, Henning Schulzrinne, 07-MAR-03,
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. Most of the text in this memorandum is identical to RFC 1889 which it obsoletes. There are no changes in the packet formats on the wire, only changes to the rules and algorithms governing how the protocol is used. The biggest change is an enhancement to the scalable timer algorithm for calculating when to send RTCP packets in order to minimize transmission in excess of the intended rate when many participants join a session simultaneously. |
| rfc1890.txt | RTP Profile for Audio and Video Conferences with Minimal Control |
| Author(s) | Audio-Video Transport Working Group, H. Schulzrinne |
| Organization | ietf |
| State | proposed standard |
| Size | 37509 bytes |
| obsoleted by | rfc3551.txt |
| Abstract | This memo describes a profile for the use of the real-time transport protocol (RTP), version 2, and the associated control protocol, RTCP, within audio and video multiparticipant conferences with minimal control. It provides interpretations of generic fields within the RTP specification suitable for audio and video conferences. In particular, this document defines a set of default mappings from payload type numbers to encodings. The document also describes how audio and video data may be carried within RTP. It defines a set of standard encodings and their names when used within RTP. However, the encoding definitions are independent of the particular transport mechanism used. The descriptions provide pointers to reference implementations and the detailed standards. This document is meant as an aid for implementors of audio, video and other real-time multimedia applications. |
| draft-ietf-avt-profile-new-13.txt
|
"RTP Profile for Audio and Video Conferences with Minimal Control",
Stephen Casner, Henning Schulzrinne, 07-MAR-03,
This document describes a profile called 'RTP/AVP' for the use of the real-time transport protocol (RTP), version 2, and the associated control protocol, RTCP, within audio and video multiparticipant conferences with minimal control. It provides interpretations of generic fields within the RTP specification suitable for audio and video conferences. In particular, this document defines a set of default mappings from payload type numbers to encodings. This document also describes how audio and video data may be carried within RTP. It defines a set of standard encodings and their names when used within RTP. The descriptions provide pointers to reference implementations and the detailed standards. This document is meant as an aid for implementors of audio, video and other real-time multimedia applications. This memorandum obsoletes RFC 1890. It is mostly backwards- compatible except for functions removed because two interoperable implementations were not found. The additions to RFC 1890 codify existing practice in the use of payload formats under this profile and include new payload formats defined since RFC 1890 was published. |
| rfc2198.txt | RTP Payload for Redundant Audio Data |
| Author(s) | C. Perkins, I. Kouvelas, O. Hodson, V. Hardman, M. Handley, J.C. Bolot, A. Vega-Garcia, S. Fosse-Parisis |
| Organization | ietf |
| State | proposed standard |
| Size | 25166 bytes |
| Abstract | This document describes a payload format for use with the real-time transport protocol (RTP), version 2, for encoding redundant audio data. The primary motivation for the scheme described herein is the development of audio conferencing tools for use with lossy packet networks such as the Internet Mbone, although this scheme is not limited to such applications. |
| rfc2833.txt | RTP Payload for DTMF Digits, Telephony Tones and Telephony Signals |
| Author(s) | H. Schulzrinne, S. Petrack |
| Organization | ietf |
| State | proposed standard |
| Size | 68786 bytes |
| Abstract | This memo describes how to carry dual-tone multifrequency (DTMF) signaling, other tone signals and telephony events in RTP packets. |
| rfc2793.txt | RTP Payload for Text Conversation |
| Author(s) | G. Hellstrom |
| Organization | ietf |
| State | proposed standard |
| Size | 20674 bytes |
| Abstract | This memo describes how to carry text conversation session contents in RTP packets. Text conversation session contents are specified in ITU-T Recommendation T.140 [1]. Text conversation is used alone or in connection to other conversational facilities such as video and voice, to form multimedia conversation services. This RTP payload description contains an optional possibility to include redundant text from already transmitted packets in order to reduce the risk of text loss caused by packet loss. The redundancy coding follows RFC 2198. |
| draft-ietf-avt-ilbc-codec-01.txt
|
"Internet Low Bit Rate Codec", Steven Andersen, 07-MAR-03,
This document specifies a speech codec suitable for robust voice communication over IP. The codec is developed by Global IP Sound (GIPS). It is designed for narrow band speech and results in a payload bit rate of 13.33 kbit/s for 30 ms frames and 15.20 kbit/s for 20 ms frames. The codec enables graceful speech quality degradation in the case of lost frames, which occurs in connection with lost or delayed IP packets. |
| draft-herlein-speex-rtp-profile-00.txt | RTP Payload Format for the Speex Codec |
| Author(s) | Greg Herlein |
| Organization | ietf |
| State | unknown |
| Date | 2003-02-25 |
| Size | 18785 bytes |
| Abstract |
| draft-duric-rtp-ilbc-02.txt
|