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HTTP Fallback for RTP Media Streams
draft-miniero-rtcweb-http-fallback-00

Document type: Expired Internet-Draft (individual)
Document stream: No stream defined
Last updated: 2013-02-08 (latest revision 2012-08-07)
Intended RFC status: Unknown
Other versions: (expired, archived): plain text, pdf, html

Stream State:No stream defined
Document shepherd: No shepherd assigned

IESG State: Expired
Responsible AD: (None)
Send notices to: No addresses provided

This Internet-Draft is no longer active. Unofficial copies of old Internet-Drafts can be found here:
http://tools.ietf.org/id/draft-miniero-rtcweb-http-fallback

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

Lorenzo Miniero <lorenzo@meetecho.com>

(Note: The e-mail addresses provided for the authors of this Internet-Draft may no longer be valid)