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RealTime Internet Peering for Telephony (RIPT) Compatibility with webRTC
draft-rosenberg-dispatch-ript-webrtc-00

Document Type Expired Internet-Draft (individual)
Expired & archived
Author Jonathan Rosenberg
Last updated 2020-08-10 (Latest revision 2020-02-07)
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This Internet-Draft is no longer active.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

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

Jonathan Rosenberg

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