diff --git a/api/client-server/v1/voip.yaml b/api/client-server/v1/voip.yaml new file mode 100644 index 00000000..5fdf1ca7 --- /dev/null +++ b/api/client-server/v1/voip.yaml @@ -0,0 +1,68 @@ +swagger: '2.0' +info: + title: "Matrix Client-Server v1 Voice over IP API" + version: "1.0.0" +host: localhost:8008 +schemes: + - https + - http +basePath: /_matrix/client/api/v1 +consumes: + - application/json +produces: + - application/json +securityDefinitions: + accessToken: + type: apiKey + description: The user_id or application service access_token + name: access_token + in: query +paths: + "/turnServer": + get: + summary: Obtain TURN server credentials. + description: |- + This API provides credentials for the client to use when initiating + calls. + security: + - accessToken: [] + responses: + 200: + description: The TURN server credentials. + examples: + application/json: |- + { + "username":"1443779631:@user:example.com", + "password":"JlKfBy1QwLrO20385QyAtEyIv0=", + "uris":[ + "turn:turn.example.com:3478?transport=udp", + "turn:10.20.30.40:3478?transport=tcp", + "turns:10.20.30.40:443?transport=tcp" + ], + "ttl":86400 + } + schema: + type: object + properties: + username: + type: string + description: |- + The username to use. + password: + type: string + description: |- + The password to use. + uris: + type: array + items: + type: string + description: A list of TURN URIs + ttl: + type: integer + description: The time-to-live in seconds + required: ["username", "password", "uris", "ttl"] + 429: + description: This request was rate-limited. + schema: + "$ref": "definitions/error.yaml" + diff --git a/specification/1-client_server_api.rst b/specification/1-client_server_api.rst index 7d8d57e4..db88bbd6 100644 --- a/specification/1-client_server_api.rst +++ b/specification/1-client_server_api.rst @@ -427,6 +427,8 @@ the complete dataset is provided in "chunk". Events ------ +.. _sect:events: + Overview ~~~~~~~~ diff --git a/specification/modules/voip_events.rst b/specification/modules/voip_events.rst index 4786ae63..a7b02538 100644 --- a/specification/modules/voip_events.rst +++ b/specification/modules/voip_events.rst @@ -1,20 +1,26 @@ Voice over IP -------------- +============= .. _module:voip: -Matrix can also be used to set up VoIP calls. This is part of the core -specification, although is at a relatively early stage. Voice (and video) over -Matrix is built on the WebRTC 1.0 standard. Call events are sent to a room, like -any other event. 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. +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}} -Message Exchange -~~~~~~~~~~~~~~~~ -A call is set up with messages exchanged as follows: +Client behaviour +---------------- + +A call is set up with message events exchanged as follows: :: @@ -41,28 +47,55 @@ Or a rejected call: Calls are negotiated according to the WebRTC specification. - Glare ~~~~~ -This specification aims to address the problem of two users calling each other -at roughly the same time and their invites crossing on the wire. It is a far -better experience for the users if their calls are connected if it is clear -that their intention is to set up a call with one another. In Matrix, calls are -to rooms rather than users (even if those rooms may only contain one other user) -so we consider calls which are to the same room. The rules for dealing with such -a situation are as follows: - - - If an invite to a room is received whilst the client is preparing to send an - 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 invite to a room is received after the client has sent an 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. + +"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. Thusly, any media stream that had been +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. +