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### Voice over IP
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This module outlines how two users in a room can set up a Voice over IP
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(VoIP) call to each other. Voice and video calls are built upon the
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WebRTC 1.0 standard. Call signalling is achieved by sending [message
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events](#events) to the room. In this version of the spec, only two-party
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communication is supported (e.g. between two peers, or between a peer
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and a multi-point conferencing unit). This means that clients MUST only
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send call events to rooms with exactly two participants.
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#### Events
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{{% event-group group_name="m.call" %}}
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#### Client behaviour
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A call is set up with message events exchanged as follows:
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```
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Caller Callee
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[Place Call]
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m.call.invite ----------->
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m.call.candidate -------->
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[..candidates..] -------->
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[Answers call]
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<--------------- m.call.answer
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[Call is active and ongoing]
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<--------------- m.call.hangup
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```
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Or a rejected call:
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```
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Caller Callee
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m.call.invite ------------>
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m.call.candidate --------->
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[..candidates..] --------->
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[Rejects call]
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<-------------- m.call.hangup
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```
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Calls are negotiated according to the WebRTC specification.
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##### Glare
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"Glare" is a problem which occurs when two users call each other at
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roughly the same time. This results in the call failing to set up as
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there already is an incoming/outgoing call. A glare resolution algorithm
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can be used to determine which call to hangup and which call to answer.
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If both clients implement the same algorithm then they will both select
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the same call and the call will be successfully connected.
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As calls are "placed" to rooms rather than users, the glare resolution
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algorithm outlined below is only considered for calls which are to the
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same room. The algorithm is as follows:
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- If an `m.call.invite` to a room is received whilst the client is
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**preparing to send** an `m.call.invite` to the same room:
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- the client should cancel its outgoing call and instead
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automatically accept the incoming call on behalf of the user.
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- If an `m.call.invite` to a room is received **after the client has
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sent** an `m.call.invite` to the same room and is waiting for a
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response:
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- the client should perform a lexicographical comparison of the
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call IDs of the two calls and use the *lesser* of the two calls,
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aborting the greater. If the incoming call is the lesser, the
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client should accept this call on behalf of the user.
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The call setup should appear seamless to the user as if they had simply
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placed a call and the other party had accepted. This means any media
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stream that had been setup for use on a call should be transferred and
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used for the call that replaces it.
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#### Server behaviour
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The homeserver MAY provide a TURN server which clients can use to
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contact the remote party. The following HTTP API endpoints will be used
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by clients in order to get information about the TURN server.
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{{% http-api spec="client-server" api="voip" %}}
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#### Security considerations
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Calls should only be placed to rooms with one other user in them. If
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they are placed to group chat rooms it is possible that another user
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will intercept and answer the call.
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