End-to-End VoIP Header Compression Using cRTP
draft-ash-e2e-crtp-hdr-compress-01
| Document | Type |
Expired Internet-Draft
(individual)
Expired & archived
|
|
|---|---|---|---|
| Authors | Bur Goode , Jim Hand | ||
| Last updated | 2003-03-06 | ||
| RFC stream | (None) | ||
| Intended RFC status | (None) | ||
| Formats | |||
| Stream | Stream state | (No stream defined) | |
| Consensus boilerplate | Unknown | ||
| RFC Editor Note | (None) | ||
| IESG | IESG state | Expired | |
| Telechat date | (None) | ||
| Responsible AD | (None) | ||
| Send notices to | (None) |
This Internet-Draft is no longer active. A copy of the expired Internet-Draft is available in these formats:
Abstract
VoIP typically uses the encapsulation voice/RTP/UDP/IP, wherein the packet header is at least 40 bytes, while the voice payload is typically no more than 30 bytes. VoIP header compression can significantly reduce the VoIP overhead through various compression mechanisms. This is important on access links where bandwidth is scarce, and can be important on backbone facilities, especially where costs are high (e.g., some global cross-sections). In this draft we propose to re-use the methods in cRTP to determine the header compression context and to use the cRTP session context ID to route a compressed packet between the ingress and egress routers.
Authors
(Note: The e-mail addresses provided for the authors of this Internet-Draft may no longer be valid.)