Requirements for Interworking WebRTC with Current SIP Deployments
draft-kaplan-rtcweb-sip-interworking-requirements-02
Document | Type |
Expired Internet-Draft
(individual)
Expired & archived
|
|
---|---|---|---|
Author | Hadriel Kaplan | ||
Last updated | 2012-05-25 (Latest revision 2011-11-22) | ||
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This Internet-Draft is no longer active.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
The IETF RTCWEB WG has been discussing how to interwork WebRTC with deployed SIP equipment and domains. Doing so may require an Interworking Function middlebox in the media-plane. This document lists some WebRTC-to-SIP use-cases, the WebRTC requirements to support such, and the complexity involved in interworking if the requirements cannot be met.
Authors
(Note: The e-mail addresses provided for the authors of this Internet-Draft may no longer be valid.)