RealTime Internet Peering for Telephony
draft-rosenbergjennings-dispatch-ript-00

Document Type Expired Internet-Draft (individual)
Authors Jonathan Rosenberg  , Cullen Jennings  , Anthony Minessale  , Jason Livingood  , Justin Uberti 
Last updated 2020-08-10 (latest revision 2020-02-07)
Replaces draft-rosenbergjennings-dispatch-ripp
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This Internet-Draft is no longer active. A copy of the expired Internet-Draft can be found at
https://www.ietf.org/archive/id/draft-rosenbergjennings-dispatch-ript-00.txt

Abstract

This document specifies the Realtime Internet Peering for Telephony (RIPT) protocol. RIPT is used to provide peering of voice and video communications between entities. These include a traditional voice trunking provider (such as a telco), and a trunking consumer (such as an enterprise PBX or contact center), or between a video conferencing endpoint deployed in an enterprise, and a video conferencing SaaS service. RIPT is an alternative to SIP, SDP and RTP for these use cases, and is designed as a web application using HTTP/3. Using HTTP/3 allows implementors to build their applications on top of cloud platforms, such as AWS, Azure and Google Cloud, all of which are heavily focused on HTTP based services. RIPT also addresses many of the challenges of traditional SIP-based peering. It supports modern techniques for load balancing, autoscaling, call-preserving failover, graceful call migrations, security by default, STIR-based caller ID, provisioning, and capabilities - all of which have been challenges with traditional SIP peering and voice trunking. Since it runs over HTTP/3, it works through NATs and firewalls with the same ease as HTTP does.

Authors

Jonathan Rosenberg (jdrosen@jdrosen.net)
Cullen Jennings (fluffy@iii.ca)
Anthony Minessale (anthm@signalwire.com)
Jason Livingood (Jason_Livingood@comcast.com)
Justin Uberti (justin@uberti.name)

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