matrix-spec/specification/modules/voip_events.rst

Voice over IP
=============

.. _module:voip:

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.

.. _message events: `sect:events`_

Events
------

{{voip_events}}

Client behaviour
----------------

A call is set up with message events exchanged as follows:

::

   Caller                    Callee
 [Place Call]
 m.call.invite ----------->
 m.call.candidate -------->
 [..candidates..] -------->
                           [Answers call]
          <--------------- m.call.answer
    [Call is active and ongoing]
          <--------------- m.call.hangup

Or a rejected call:

::

   Caller                      Callee
 m.call.invite ------------>
 m.call.candidate --------->
 [..candidates..] --------->
                            [Rejects call]
            <-------------- m.call.hangup

Calls are negotiated according to the WebRTC specification.

Glare
~~~~~

"Glare" is a problem which occurs when two users call each other at roughly the
same time. This results in the call failing to set up as there already is an
incoming/outgoing call. A glare resolution algorithm can be used to determine
which call to hangup and which call to answer. If both clients implement the
same algorithm then they will both select the same call and the call will be
successfully connected.


As calls are "placed" to rooms rather than users, the glare resolution algorithm
outlined below is only considered for calls which are to the same room. The
algorithm is as follows:

- If an ``m.call.invite`` to a room is received whilst the client is
  **preparing to send** an ``m.call.invite`` to the same room:

  * the client should cancel its outgoing call and instead
    automatically accept the incoming call on behalf of the user.

- If an ``m.call.invite`` to a room is received **after the client has sent**
  an ``m.call.invite`` to the same room and is waiting for a response:

  * the client should perform a lexicographical comparison of the call IDs of
    the two calls and use the *lesser* of the two calls, aborting the
    greater. If the incoming call is the lesser, the client should accept
    this call on behalf of the user.


The call setup should appear seamless to the user as if they had simply placed
a call and the other party had accepted. This means any media stream that had been
setup for use on a call should be transferred and used for the call that
replaces it.

Server behaviour
----------------

The homeserver MAY provide a TURN server which clients can use to contact the
remote party. The following HTTP API endpoints will be used by clients in order
to get information about the TURN server.

{{voip_http_api}}


Security considerations
-----------------------

Calls should only be placed to rooms with one other user in them. If they are
placed to group chat rooms it is possible that another user will intercept and
answer the call.