RealTime Internet Peering for Telephony (RIPT) Compatibility with webRTC
draft-rosenberg-dispatch-ript-webrtc-00
| Document | Type | Expired Internet-Draft (individual) | |
|---|---|---|---|
| Author | Jonathan Rosenberg | ||
| Last updated | 2020-08-10 (Latest revision 2020-02-07) | ||
| Stream | (None) | ||
| Intended RFC status | (None) | ||
| Formats |
Expired & archived
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| 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) |
https://www.ietf.org/archive/id/draft-rosenberg-dispatch-ript-webrtc-00.txt
Abstract
The Real-Time Internet Peering for Telephony (RIPT) Protocol defines a technique for establishing, terminating and otherwise managing calls between entities in differing administrative domains. The RIPT Inbound extension brings this to end clients, such as a browser. However, it defines a different technique for media that cannot directly use the webRTC APIs, and require a change to them. This specification provides an extension to RIPT for webRTC compatibility, enabling media to flow from browser to server as is done with RIPT, or from browser to browser as is done with webRTC. It also discusses techniques for sending e2e encrypted media.
Authors
(Note: The e-mail addresses provided for the authors of this Internet-Draft may no longer be valid.)