Add spec for MSC2746 (#1511)

* Change version field to a string

And add the notes on how the version field works.

* Add spec requiring tracks to be within streams.

* Put streams spec in its own section

* Add 'invitee' field

* Add party_id

* Remember how JSON works

* Add m.call.select_answer

* Update examples

* Add select_answer to call flow example diagram

* Add m.call.reject

* Make party_id required in other events

* Add possible ways for client to handle an invite

* Convert hangup & reject events to YAML

So we can have a bulleted list in the description for the values
of 'reason'.

* Add new reason codes to hangup & reject

* Add m.call.negotiate

* Add other sections

* Revert changes to package lock

* Typos

* Fix type of other version fields, fix anchor.

* Add newsfragment

* Fix reason in hangup/reject

* Change tense

Co-authored-by: Hubert Chathi <hubertc@matrix.org>

* Tense, typos & grammar

Co-authored-by: Hubert Chathi <hubertc@matrix.org>

* Linkify

Co-authored-by: Hubert Chathi <hubertc@matrix.org>

* Remove unnecessary parts from link

Co-authored-by: Hubert Chathi <hubertc@matrix.org>

* Capitalise

Co-authored-by: Hubert Chathi <hubertc@matrix.org>

* Fix hangup reasons

* Clarify who can answer

Co-authored-by: Hubert Chathi <hubertc@matrix.org>

* Linkify

Co-authored-by: Hubert Chathi <hubertc@matrix.org>

* Remove reference to 'this MSC'.

* Move common VoIP fields into a call event type.

* Move common voip events to the content, not the actual event

* Remove reason from reject event

I confused myself, but it's not in the MSC and it shouldn't be.

* Failure to YAML

* Fix number of room members allowed when sending voip events.

Co-authored-by: Hubert Chathi <hubertc@matrix.org>

* Add 'added in' version

Co-authored-by: Hubert Chathi <hubertc@matrix.org>

* Another added-in

Co-authored-by: Hubert Chathi <hubertc@matrix.org>

* Add missing comma

---------

Co-authored-by: Hubert Chathi <hubertc@matrix.org>
Co-authored-by: Travis Ralston <travisr@matrix.org>

changelogs/client_server/newsfragments/1511.feature

@@ -0,0 +1 @@
+Update VoIP spec for [MSC2746](https://github.com/matrix-org/matrix-spec-proposals/pull/2746).

content/client-server-api/modules/voip_events.md

@@ -6,11 +6,128 @@ This module outlines how two users in a room can set up a Voice over IP
 WebRTC 1.0 standard. Call signalling is achieved by sending [message
 events](#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.
+and a multi-point conferencing unit). Calls can take place in rooms with
+multiple members, but only two devices can take part in the call.
+
+All VoIP events have a `version` field. This is used to determine whether
+devices support this new version of the protocol. For example, clients can use
+this field to know whether to expect an `m.call.select_answer` event from their
+opponent. If clients see events with `version` other than `0` or `"1"`
+(including, for example, the numeric value `1`), they should treat these the
+same as if they had `version` == `"1"`.
+
+Note that this implies any and all future versions of VoIP events should be
+backwards-compatible.  If it does become necessary to introduce a non
+backwards-compatible VoIP spec, the intention would be for it to simply use a
+separate set of event types.
+
+#### Party Identifiers
+Whenever a client first participates in a new call, it generates a `party_id` for itself to use for the
+duration of the call. This needs to be long enough that the chance of a collision between multiple devices
+both generating an answer at the same time generating the same party ID is vanishingly small: 8 uppercase +
+lowercase alphanumeric characters is recommended. Parties in the call are identified by the tuple of
+`(user_id, party_id)`.
+
+The client  adds a `party_id` field containing this ID to the top-level of the content of all VoIP events
+it sends on the call, including `m.call.invite`. Clients use this to identify remote echo of their own
+events: since a user may call themselves, they cannot simply ignore events from their own user. This
+field also identifies different answers sent by different clients to an invite, and matches `m.call.candidates`
+events to their respective answer/invite.
+
+A client implementation may choose to use the device ID used in end-to-end cryptography for this purpose,
+or it may choose, for example, to use a different one for each call to avoid leaking information on which
+devices were used in a call (in an unencrypted room) or if a single device (ie. access token) were used to
+send signalling for more than one call party.
+
+A grammar for `party_id` is defined [below](#grammar-for-voip-ids).
+
+#### Politeness
+In line with [WebRTC perfect negotiation](https://w3c.github.io/webrtc-pc/#perfect-negotiation-example)
+there are rules to establish which party is polite in the process of renegotiation. The callee is
+always the polite party. In a glare situation, the politenes of a party is therefore determined by
+whether the inbound or outbound call is used: if a client discards its outbound call in favour of
+an inbound call, it becomes the polite party.
+
+#### Call Event Liveness
+`m.call.invite` contains a `lifetime` field that indicates how long the offer is valid for. When
+a client receives an invite, it should use the event's `age` field in the sync response plus the
+time since it received the event from the homeserver to determine whether the invite is still valid.
+The use of the `age` field ensures that incorrect clocks on client devices don't break calls.
+
+If the invite is still valid *and will remain valid for long enough for the user to accept the call*,
+it should signal an incoming call. The amount of time allowed for the user to accept the call may
+vary between clients. For example, it may be longer on a locked mobile device than on an unlocked
+desktop device.
+
+The client should only signal an incoming call in a given room once it has completed processing the
+entire sync response and, for encrypted rooms, attempted to decrypt all encrypted events in the
+sync response for that room. This ensures that if the sync response contains subsequent events that
+indicate the call has been hung up, rejected, or answered elsewhere, the client does not signal it.
+
+If on startup, after processing locally stored events, the client determines that there is an invite
+that is still valid, it should still signal it but only after it has completed a sync from the homeserver.
+
+The minimal recommended lifetime is 90 seconds - this should give the user enough time to actually pick
+up the call.
+
+#### ICE Candidate Batching
+Clients should aim to send a small number of candidate events, with guidelines:
+ * ICE candidates which can be discovered immediately or almost immediately in the invite/answer
+   event itself (eg. host candidates). If server reflexive or relay candidates can be gathered in
+   a sufficiently short period of time, these should be sent here too. A delay of around 200ms is
+   suggested as a starting point.
+ * The client should then allow some time for further candidates to be gathered in order to batch them,
+   rather than sending each candidate as it arrives. A starting point of 2 seconds after sending the
+   invite or 500ms after sending the answer is suggested as a starting point (since a delay is natural
+   anyway after the invite whilst the client waits for the user to accept it).
+
+#### End-of-candidates
+An ICE candidate whose value is the empty string means that no more ICE candidates will
+be sent. Clients must send such a candidate in an `m.call.candidates` message.
+The WebRTC spec requires browsers to generate such a candidate, however note that at time of writing,
+not all browsers do (Chrome does not, but does generate an `icegatheringstatechange` event). The
+client should send any remaining candidates once candidate generation finishes, ignoring timeouts above.
+This allows bridges to batch the candidates together when bridging to protocols that don't support
+trickle ICE.
+
+#### DTMF
+Matrix clients can send DTMF as specified by WebRTC. The WebRTC standard as of August
+2020 does not support receiving DTMF but a Matrix client can receive and interpret the DTMF sent
+in the RTP payload.
+
+#### Grammar for VoIP IDs
+`call_id`s and `party_id` are explicitly defined to be between 1 and 255 characters long, consisting
+of the characters `[0-9a-zA-Z._~-]`.
+
+(Note that this matches the grammar of 'opaque IDs' from
+[MSC1597](https://github.com/matrix-org/matrix-spec-proposals/blob/rav/proposals/id_grammar/proposals/1597-id-grammar.md#opaque-ids),
+and that of the `id` property of the
+ [`m.login.sso` flow schema](#definition-mloginsso-flow-schema).)
+
+#### Behaviour on Room Leave
+If the client sees the user it is in a call with leave the room, the client should treat this
+as a hangup event for any calls that are in progress. No specific requirement is given for the
+situation where a client has sent an invite and the invitee leaves the room, but the client may
+wish to treat it as a rejection if there are no more users in the room who could answer the call
+(eg. the user is now alone or the `invitee` field was set on the invite).
+
+The same behaviour applies when a client is looking at historic calls.
+
+#### Supported Codecs
+The Matrix spec does not mandate particular audio or video codecs, but instead defers to the
+WebRTC spec. A compliant Matrix VoIP client will behave in the same way as a supported 'browser'
+in terms of what codecs it supports and what variants thereof. The latest WebRTC specification
+applies, so clients should keep up to date with new versions of the WebRTC specification whether
+or not there have been any changes to the Matrix spec.
 
 #### Events
 
+##### Common Fields
+
+{{% event-fields event_type="call_event" %}}
+
+##### Events
+
 {{% event-group group_name="m.call" %}}
 
 #### Client behaviour
@@ -25,6 +142,7 @@ A call is set up with message events exchanged as follows:
     [..candidates..] -------->
                             [Answers call]
            <--------------- m.call.answer
+    m.call.select_answer ----------->
      [Call is active and ongoing]
            <--------------- m.call.hangup
 ```
@@ -42,6 +160,43 @@ Or a rejected call:
 
 Calls are negotiated according to the WebRTC specification.
 
+In response to an incoming invite, a client may do one of several things:
+ * Attempt to accept the call by sending an `m.call.answer`.
+ * Actively reject the call everywhere: send an `m.call.reject` as per above, which will stop the call from
+   ringing on all the user's devices and the caller's client will inform them that the user has
+   rejected their call.
+ * Ignore the call: send no events, but stop alerting the user about the call. The user's other
+   devices will continue to ring, and the caller's device will continue to indicate that the call
+   is ringing, and will time the call out in the normal way if no other device responds.
+
+##### Streams
+
+Clients are expected to send one stream with one track of kind `audio` (creating a
+voice call). They can optionally send a second track in the same stream of kind
+`video` (creating a video call).
+
+Clients implementing this specification use the first stream and will ignore
+any streamless tracks. Note that in the JavaScript WebRTC API, this means
+`addTrack()` must be passed two parameters: a track and a stream, not just a
+track, and in a video call the stream must be the same for both audio and video
+track.
+
+A client may send other streams and tracks but the behaviour of the other party
+with respect to presenting such streams and tracks is undefined.
+
+##### Invitees
+The `invitee` field should be added whenever the call is intended for one
+specific user, and should be set to the Matrix user ID of that user. Invites
+without an `invitee` field are defined to be intended for any member of the
+room other than the sender of the event.
+
+Clients should consider an incoming call if they see a non-expired invite event where the `invitee` field is either
+absent or equal to their user's Matrix ID, however they should evaluate whether or not to ring based on their
+user's trust relationship with the callers and/or where the call was placed. As a starting point, it is
+suggested that clients ignore call invites from users in public rooms. It is strongly recommended that
+when clients do not ring for an incoming call invite, they still display the call invite in the room and
+annotate that it was ignored.
+
 ##### Glare
 
 "Glare" is a problem which occurs when two users call each other at

data/event-schemas/examples/m.call.answer.yaml

@@ -2,7 +2,8 @@
   "$ref": "core/room_event.json",
   "type": "m.call.answer",
   "content": {
-    "version" : 0,
+    "version" : "1",
+    "party_id": "67890",
     "call_id": "12345",
     "answer": {
       "type" : "answer",

data/event-schemas/examples/m.call.candidates.yaml

@@ -2,7 +2,8 @@
   "$ref": "core/room_event.json",
   "type": "m.call.candidates",
   "content": {
-    "version" : 0,
+    "version" : "1",
+    "party_id": "67890",
     "call_id": "12345",
     "candidates": [
       {

data/event-schemas/examples/m.call.hangup.yaml

@@ -2,7 +2,9 @@
   "$ref": "core/room_event.json",
   "type": "m.call.hangup",
   "content": {
-    "version" : 0,
-    "call_id": "12345"
+    "version" : "1",
+    "party_id": "67890",
+    "call_id": "12345",
+    "reason": "user_hangup"
   }
 }

data/event-schemas/examples/m.call.invite.yaml

@@ -2,7 +2,8 @@
   "$ref": "core/room_event.json",
   "type": "m.call.invite",
   "content": {
-    "version" : 0,
+    "version" : "1",
+    "party_id": "67890",
     "call_id": "12345",
     "lifetime": 60000,
     "offer": {

data/event-schemas/examples/m.call.negotiate.yaml

@@ -0,0 +1,14 @@
+{
+  "$ref": "core/room_event.json",
+  "type": "m.call.negotiate",
+  "content": {
+    "version" : "1",
+    "party_id": "67890",
+    "call_id": "12345",
+    "lifetime": 10000,
+    "offer": {
+      "type" : "offer",
+      "sdp" : "v=0\r\no=- 6584580628695956864 2 IN IP4 127.0.0.1[...]"
+    }
+  }
+}

data/event-schemas/examples/m.call.reject.yaml

@@ -0,0 +1,9 @@
+{
+  "$ref": "core/room_event.json",
+  "type": "m.call.reject",
+  "content": {
+    "version" : "1",
+    "party_id": "67890",
+    "call_id": "12345"
+  }
+}

data/event-schemas/examples/m.call.select_answer.yaml

@@ -0,0 +1,10 @@
+{
+  "$ref": "core/room_event.json",
+  "type": "m.call.select_answer",
+  "content": {
+    "version" : "1",
+    "call_id": "12345",
+    "party_id": "67890",
+    "selected_party_id": "111213"
+  }
+}

data/event-schemas/schema/core-event-schema/call_event.yaml

@@ -0,0 +1,25 @@
+description: "The content of all call events shares a set of common fields: those
+  of room events and some additional VoIP specific fields."
+properties:
+  call_id:
+    type: string
+    description: The ID of the call this event relates to.
+  version:
+    type: string
+    description: The version of the VoIP specification this message adheres to.
+      This specification is version 1. This field is a string such that experimental
+      implementations can use non-integer versions. This field was an integer
+      in the previous spec version and implementations must accept an integer
+      0.
+  party_id:
+    type: string
+    description: 'This identifies the party that sent this event. A client may
+      choose to re-use the device ID from end-to-end cryptography for the value
+      of this field.'
+    x-addedInMatrixVersion: "1.7"
+required:
+- call_id
+- version
+- party_id
+title: CallEvent
+type: object

data/event-schemas/schema/m.call.answer.yaml

@@ -7,11 +7,10 @@
     "properties": {
         "content": {
             "type": "object",
+            "allOf": [{
+                "$ref": "core-event-schema/call_event.yaml"
+            }],
             "properties": {
-                "call_id": {
-                    "type": "string",
-                    "description": "The ID of the call this event relates to."
-                },
                 "answer": {
                     "type": "object",
                     "title": "Answer",
@@ -28,13 +27,9 @@
                         }
                     },
                     "required": ["type", "sdp"]
-                },
-                "version": {
-                    "type": "number",
-                    "description": "The version of the VoIP specification this messages adheres to. This specification is version 0."
                 }
             },
-            "required": ["call_id", "answer", "version"]
+            "required": ["answer"]
         },
         "type": {
             "type": "string",

data/event-schemas/schema/m.call.candidates.yaml

@@ -7,11 +7,10 @@
     "properties": {
         "content": {
             "type": "object",
+            "allOf": [{
+                "$ref": "core-event-schema/call_event.yaml"
+            }],
             "properties": {
-                "call_id": {
-                    "type": "string",
-                    "description": "The ID of the call this event relates to."
-                },
                 "candidates": {
                     "type": "array",
                     "description": "Array of objects describing the candidates.",
@@ -34,13 +33,9 @@
                         },
                         "required": ["candidate", "sdpMLineIndex", "sdpMid"]
                     }
-                },
-                "version": {
-                    "type": "integer",
-                    "description": "The version of the VoIP specification this messages adheres to. This specification is version 0."
                 }
             },
-            "required": ["call_id", "candidates", "version"]
+            "required": ["candidates"]
         },
         "type": {
             "type": "string",