From 6925547875d0685a5ad49ec37930527236461d93 Mon Sep 17 00:00:00 2001 From: Matthew Hodgson Date: Thu, 11 Feb 2016 18:34:28 +0000 Subject: [PATCH] remove the oversimplification that the WebRTC standard is based around two-oparty communication --- specification/modules/voip_events.rst | 9 +++++---- 1 file changed, 5 insertions(+), 4 deletions(-) diff --git a/specification/modules/voip_events.rst b/specification/modules/voip_events.rst index 6945a09e2..348aad620 100644 --- a/specification/modules/voip_events.rst +++ b/specification/modules/voip_events.rst @@ -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`_