mirror of
https://github.com/livekit/livekit.git
synced 2026-06-07 08:51:56 +00:00
David Zhao
66 lines
2.4 KiB
Markdown
66 lines
2.4 KiB
Markdown
# livekit-server
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## Building
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Ensure Go 1.14+ is installed, and GOPATH/bin is in your PATH.
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Run `make proto && make
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## CLI
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A CLI is provided to make debugging & testing easier. One of the utilities is the ability to publish tracks from static files.
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To use the publish client, download the following files to your computer:
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* [happier.ivf](https://www.dropbox.com/s/4ze93d6070s0qj7/happier.ivf?dl=0) - audio track in VP8
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* [happier.ogg](https://www.dropbox.com/s/istrnolnh7avftq/happier.ogg?dl=0) - audio track in ogg
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To run a peer publishing to a room, do the following:
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1. Ensure server is running in dev mode
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```
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./bin/livekit-server --dev
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```
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2. Create a room
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```
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./bin/livekit-cli create-room
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```
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It'll print out something like this. note the room id
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```json
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{
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"sid": "RM_CkjigXb6oZQyZ4JNFZqBen",
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"node_ip": "98.35.19.21",
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"creation_time": 1607240104,
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"token": "b9e8c9f6-fbb3-46d5-b6fc-5517186510a6"
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}
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```
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3. Join room as publishing client
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```
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./ bin/livekit-cli join --audio <path/to/ogg> --video <path/to/ivf> --room-id <room_sid>
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```
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That's it, join the room with another peer id and see it receiving those tracks
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## Protocol
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LiveKit provides room based audio/video/data channels based on WebRTC.
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It provides a set of APIs to manipulate rooms, as well as its own signaling protocol to exchange room and participant information.
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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).
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The RTC service provides the signaling and everything else when the client interacts with the room. RTC service requires bidirectional
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communication between the client and server, and exchanges messages via WebSocket. Messages are encoded in either JSON or binary protobuf,
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see `rtc.proto` for the message structure.
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The flow for interaction is:
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1. Establish WebSocket to ws://<host>:<port>/rtc
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1. Server will send back a `SignalResponse` with a `join` response. It'll include the new participant's details, and what other participants are in the room
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1. Client sends a `SignalRequest` with an WebRTC `offer`
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1. Server will send back a `SignalResponse` with an `answer`
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1. Client and server will exchange ice candidates via `trickle` in the request & responses
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