HTTP Fallback for RTP Media Streams
draft-miniero-rtcweb-http-fallback-00
| Document | Type |
Expired Internet-Draft
(individual)
Expired & archived
|
|
|---|---|---|---|
| Author | Lorenzo Miniero | ||
| Last updated | 2013-02-08 (Latest revision 2012-08-07) | ||
| 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This Internet-Draft is no longer active.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
Almost all VoIP endpoints, especially SIP and RTCWEB ones, make use of RTP to tranport media frames in real-time and communicate with each other. Since RTP uses UDP, the presence of network elements that filter UDP packets and/or only allow some protocols like SMTP or HTTP to pass through would make such a communication very hard to accomplish, if not impossible. This draft describes a way to implement an HTTP Fallback for RTP media streams, that is, a way to effectively encapsulate RTP packets in HTTP messages in order to traverse proxies and firewalls.
Authors
(Note: The e-mail addresses provided for the authors of this Internet-Draft may no longer be valid.)