mirror of
https://github.com/livekit/livekit.git
synced 2026-06-06 15:31:51 +00:00
David Zhao
55 lines
2.1 KiB
Markdown
55 lines
2.1 KiB
Markdown
# livekit-server
|
|
|
|
## Building
|
|
|
|
Ensure Go 1.14+ is installed, and GOPATH/bin is in your PATH.
|
|
|
|
Run `make proto && make
|
|
|
|
## CLI
|
|
|
|
A CLI is provided to make debugging & testing easier. One of the utilities is the ability to publish tracks from static files.
|
|
|
|
To use the publish client, download the following files to your computer:
|
|
* [happier.ivf](https://www.dropbox.com/s/4ze93d6070s0qj7/happier.ivf?dl=0) - audio track in VP8
|
|
* [happier.ogg](https://www.dropbox.com/s/istrnolnh7avftq/happier.ogg?dl=0) - audio track in ogg
|
|
|
|
To run a peer publishing to a room, do the following:
|
|
|
|
1. Ensure server is running in dev mode
|
|
```
|
|
./bin/livekit-server --dev
|
|
```
|
|
|
|
2. Create a room
|
|
|
|
```
|
|
./bin/livekit-cli create-room --room-id hello
|
|
```
|
|
|
|
3. Join room as publishing client
|
|
|
|
```
|
|
./ bin/livekit-cli join --room-id hello --audio <path/to/ogg> --video <path/to/ivf>
|
|
```
|
|
|
|
That's it, join the room with another peer id and see it receiving those tracks
|
|
|
|
## Protocol
|
|
|
|
LiveKit provides room based audio/video/data channels based on WebRTC.
|
|
It provides a set of APIs to manipulate rooms, as well as its own signaling protocol to exchange room and participant information.
|
|
|
|
Room APIs are defined in room.proto, it's fairly straight forward with typical CRUD APIs. Room APIs are HTTP, built with Twirp and follows [its the conventions](https://twitchtv.github.io/twirp/docs/routing.html).
|
|
|
|
The RTC service provides the signaling and everything else when the client interacts with the room. RTC service requires bidirectional
|
|
communication between the client and server, and exchanges messages via WebSocket. Messages are encoded in either JSON or binary protobuf,
|
|
see `rtc.proto` for the message structure.
|
|
|
|
The flow for interaction is:
|
|
1. Establish WebSocket to ws://<host>:<port>/rtc
|
|
1. Server will send back a `SignalResponse` with a `join` response. It'll include the new participant's details (and in the future, room info)
|
|
1. Client sends a `SignalRequest` with an WebRTC `offer`
|
|
1. Server will send back a `SignalResponse` with an `answer`
|
|
1. Client and server will exchange ice candidates via `trickle` in the request & responses
|