You cannot select more than 25 topics
Topics must start with a letter or number, can include dashes ('-') and can be up to 35 characters long.
103 lines
3.2 KiB
ReStructuredText
103 lines
3.2 KiB
ReStructuredText
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. 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`_
|
|
|
|
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.
|
|
|