Real Time Internet Peering for Telephony (RIPT) Comparison with the Session Initiaton Protocol (SIP)
draft-rosenberg-dispatch-ript-sipdiffs-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This Internet-Draft is no longer active.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 and its extension for inbound calls to single user devices provide an alternative to the Session Initiation Protocol (SIP) for several use cases. This leads to many questions - how is RIPT different from SIP and why? What should be standardized and what should not? How much of SIP do those two specifications replace? This document discusses the differences and their motivations, and presents an analysis across the set of SIP specifications, and analyzes whether the two RIPT documents replace each with similar capability, whether they eliminate the need for that specification, or whether some or all of that specification are not addressed by RIPT.
Authors
(Note: The e-mail addresses provided for the authors of this Internet-Draft may no longer be valid.)