mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-12-21 15:56:42 +00:00
97 lines
3.5 KiB
Markdown
97 lines
3.5 KiB
Markdown
## Terminology
|
|
|
|
### Client
|
|
|
|
A GStreamer-based application
|
|
|
|
### Browser
|
|
|
|
A JS application that runs in the browser and uses built-in browser webrtc APIs
|
|
|
|
### Peer
|
|
|
|
Any webrtc-using application that can participate in a call
|
|
|
|
### Signalling server
|
|
|
|
Basic websockets server implemented in Python that manages the peers list and shovels data between peers
|
|
|
|
## Overview
|
|
|
|
This is a basic protocol for doing 1-1 audio+video calls between a gstreamer app and a JS app in a browser.
|
|
|
|
## Peer registration
|
|
|
|
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.
|
|
|
|
This protocol builds upon https://github.com/shanet/WebRTC-Example/
|
|
|
|
* Connect to the websocket server
|
|
* Send `HELLO <uid>` where `<uid>` is a string which will uniquely identify this peer
|
|
* Receive `HELLO`
|
|
* Any other message starting with `ERROR` is an error.
|
|
|
|
### 1-1 calls with a 'session'
|
|
|
|
* To connect to a single peer, send `SESSION <uid>` where `<uid>` identifies the peer to connect to, and receive `SESSION_OK`
|
|
* All further messages will be forwarded to the peer
|
|
* The call negotiation with the peer can be started by sending JSON encoded SDP (the offer) and ICE
|
|
* 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)
|
|
|
|
* Closure of the server connection means the call has ended; either because the other peer ended it or went away
|
|
* To end the call, disconnect from the server. You may reconnect again whenever you wish.
|
|
|
|
### Multi-party calls with a 'room'
|
|
|
|
* 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
|
|
* 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
|
|
* To send messages to a specific peer within the room for call negotiation (or any other purpose, use `ROOM_PEER_MSG <peer_id> <msg>`
|
|
* When a new peer joins the room, you will receive a `ROOM_PEER_JOINED <peer_id>` message
|
|
- For the purposes of convention and to avoid overwhelming newly-joined peers, offers must only be sent by the newly-joined peer
|
|
* When a peer leaves the room, you will receive a `ROOM_PEER_LEFT <peer_id>` message
|
|
- You should stop sending/receiving media from/to this peer
|
|
* To get a list of all peers currently in the room, send `ROOM_PEER_LIST` and receive `ROOM_PEER_LIST <peer1_id> ...`
|
|
- This list will never contain your own `<uid>`
|
|
- 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
|
|
* You may stay connected to a room for as long as you like
|
|
|
|
## Negotiation
|
|
|
|
Once a call has been setup with the signalling server, the peers must negotiate SDP and ICE candidates with each other.
|
|
|
|
The calling side must create an SDP offer and send it to the peer as a JSON object:
|
|
|
|
```json
|
|
{
|
|
"sdp": {
|
|
"sdp": "o=- [....]",
|
|
"type": "offer"
|
|
}
|
|
}
|
|
```
|
|
|
|
The callee must then reply with an answer:
|
|
|
|
```json
|
|
{
|
|
"sdp": {
|
|
"sdp": "o=- [....]",
|
|
"type": "answer"
|
|
}
|
|
}
|
|
```
|
|
|
|
ICE candidates must be exchanged similarly by exchanging JSON objects:
|
|
|
|
|
|
```json
|
|
{
|
|
"ice": {
|
|
"candidate": ...,
|
|
"sdpMLineIndex": ...,
|
|
...
|
|
}
|
|
}
|
|
```
|
|
|
|
Note that the structure of these is the same as that specified by the WebRTC spec.
|