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97 lines
3.5 KiB
Markdown
97 lines
3.5 KiB
Markdown
## Terminology
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### Client
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A GStreamer-based application
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### Browser
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A JS application that runs in the browser and uses built-in browser webrtc APIs
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### Peer
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Any webrtc-using application that can participate in a call
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### Signalling server
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Basic websockets server implemented in Python that manages the peers list and shovels data between peers
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## Overview
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This is a basic protocol for doing 1-1 audio+video calls between a gstreamer app and a JS app in a browser.
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## Peer registration
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Peers must register with the signalling server before a call can be initiated. The server connection should stay open as long as the peer is available or in a call.
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This protocol builds upon https://github.com/shanet/WebRTC-Example/
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* Connect to the websocket server
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* Send `HELLO <uid>` where `<uid>` is a string which will uniquely identify this peer
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* Receive `HELLO`
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* Any other message starting with `ERROR` is an error.
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### 1-1 calls with a 'session'
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* To connect to a single peer, send `SESSION <uid>` where `<uid>` identifies the peer to connect to, and receive `SESSION_OK`
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* All further messages will be forwarded to the peer
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* The call negotiation with the peer can be started by sending JSON encoded SDP (the offer) and ICE
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* You can also ask the peer to send the SDP offer and begin sending ICE candidates. After `SESSION_OK` if you send `OFFER_REQUEST`, the peer will take over. (NEW in 1.19, not all clients support this)
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* Closure of the server connection means the call has ended; either because the other peer ended it or went away
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* To end the call, disconnect from the server. You may reconnect again whenever you wish.
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### Multi-party calls with a 'room'
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* To create a multi-party call, you must first register (or join) a room. Send `ROOM <room_id>` where `<room_id>` is a unique room name
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* Receive `ROOM_OK ` from the server if this is a new room, or `ROOM_OK <peer1_id> <peer2_id> ...` where `<peerN_id>` are unique identifiers for the peers already in the room
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* To send messages to a specific peer within the room for call negotiation (or any other purpose, use `ROOM_PEER_MSG <peer_id> <msg>`
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* When a new peer joins the room, you will receive a `ROOM_PEER_JOINED <peer_id>` message
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- For the purposes of convention and to avoid overwhelming newly-joined peers, offers must only be sent by the newly-joined peer
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* When a peer leaves the room, you will receive a `ROOM_PEER_LEFT <peer_id>` message
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- You should stop sending/receiving media from/to this peer
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* To get a list of all peers currently in the room, send `ROOM_PEER_LIST` and receive `ROOM_PEER_LIST <peer1_id> ...`
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- This list will never contain your own `<uid>`
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- In theory you should never need to use this since you are guaranteed to receive JOINED and LEFT messages for all peers in a room
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* You may stay connected to a room for as long as you like
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## Negotiation
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Once a call has been setup with the signalling server, the peers must negotiate SDP and ICE candidates with each other.
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The calling side must create an SDP offer and send it to the peer as a JSON object:
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```json
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{
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"sdp": {
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"sdp": "o=- [....]",
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"type": "offer"
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}
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}
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```
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The callee must then reply with an answer:
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```json
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{
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"sdp": {
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"sdp": "o=- [....]",
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"type": "answer"
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}
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}
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```
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ICE candidates must be exchanged similarly by exchanging JSON objects:
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```json
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{
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"ice": {
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"candidate": ...,
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"sdpMLineIndex": ...,
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...
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}
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}
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```
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Note that the structure of these is the same as that specified by the WebRTC spec.
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