charter.txt

This working group will standardize a protocol, capable of operating
atop HTTP/3, which supports real-time voice and video communications,
including signaling, media negotiation, and transmission of media
packets.

The primary rationale for this new protocol is to enable deployment of
real-time communications services on modern "cloud" systems. These
systems are built around HTTP, and for HTTP-based applications, they
enable load balancing, HA, auto-scaling, and so on. However, real-time
communications protocols are based on SIP and RTP, and they cannot take
advantage of these HTTP-based capabilities.

It is a non-goal of this working group to replicate all of SIP and its
many extensions into HTTP. The group will limit itself to supporting the
functionality in widespread actual usage today.

The protocol uses HTTP techniques for authentication and authorization
(notably OAuth), and requires hop by hop encryption (i.e., https). The
protocol will also allow for e2e media encryption, although keying is
out of scope, and is expected to be handled by other protocols such as
MLS. This extension will also utilize STIR for callerID.

This protocol should be implementable in browsers, thick desktop
clients, mobile apps and servers.

The group will standardize an extension to the baseline protocol that
automates the configuration necessary to achieve calling between on
organization which is a customer of the other (for example, enterprise
to service provider).

[OPEN ISSUE: Is P2P media in or out? If in, ICE or something else ?]

The group will do its work in conjunction with active software
development efforts, so that implementation experience feeds directly
into protocol development.

The group will coordinate with the Webtranport, QUIC, HTTPbis and STIR
working groups.

Milestones:

Dec 2020: Submit baseline protocol to IESG
Mar 2021: Submit autoconfiguration protocol to IESG