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remove the oversimplification that the WebRTC standard is based around two-oparty communication

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Matthew Hodgson 2016-02-11 18:34:28 +00:00
parent 41130314da
commit 6925547875
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9
specification/modules/voip_events.rst
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@ -5,10 +5,11 @@ Voice over IP
This module outlines how two users in a room can set up a Voice over IP (VoIP)
call to each other. Voice and video calls are built upon the WebRTC 1.0 standard.
Call signalling is achieved by sending `message events`_ to the room. As a result,
this means that clients MUST only send call events to rooms with exactly two
participants as currently the WebRTC standard is based around two-party
communication.
Call signalling is achieved by sending `message events`_ to the room. In this
version of the spec, only two-party communication is supported (e.g. between two
peers, or between a peer and a multi-point conferencing unit).
This means that clients MUST only send call events to rooms with exactly two
participants.
.. _message events: `sect:events`_