LiveKit leverages WebRTC technology, providing developers with a seamless framework for integrating real-time features into their applications. In my previous article we discussed how Websockets works link. In this article we explores how LiveKit handles calls, from signaling to media routing.
Components of a LiveKit Call
1. Signaling: Used for exchanging metadata, such as connection details and media capabilities, between participants.
2. Selective Forwarding Unit (SFU): Livekit server routes media streams efficiently, ensuring minimal latency and bandwidth optimization.
3. STUN/TURN Server: Helps relay media streams in cases where client is behind NAT, which is usually the scenario.
4. Client SDKs: Provides APIs for applications to connect to LiveKit rooms and manage audio, video, and data streams.
Step-by-Step: How a Call is Handled
1. Joining a Room
When a user initiates a call, their client application connects to a LiveKit room. This process begins with signaling over a WebSocket connection.
• The client sends a JoinRequest to the LiveKit server, including a JSON Web Token (JWT) for authentication.
• The server responds with a JoinResponse, providing details about the room and existing participants.
2. WebRTC Negotiation
Once the client joins, WebRTC negotiation begins to establish media connections.
• The server sends an SDP (Session Description Protocol) offer to the client, describing media capabilities and codecs. The client processes the offer and replies with an SDP answer.
3. Publishing Media Tracks
Participants can publish their audio, video, or data streams to the room. LiveKit uses the SFU to manage these tracks.
The SFU forwards published tracks selectively to subscribers, optimizing bandwidth.
4. Subscribing to Media Tracks
When new participants join or new tracks are published, existing clients receive notifications to subscribe to these tracks.
Clients can choose which tracks to subscribe to using selective subscription, reducing bandwidth usage.
5. Handling NAT Traversal
To ensure connectivity across different network environments, LiveKit uses a combination of STUN and TURN servers.
• STUN helps discover public IP and port information for direct connections.
• TURN relays media when direct paths are blocked, ensuring robust connectivity.
6. Real-Time Communication
With the media streams established, participants can now engage in real-time communication.
Livekit Main Features
1. Efficient Media Routing: The SFU ensures that only the necessary streams are forwarded to each participant, reducing bandwidth requirements.
2. Dynamic Adaptation: LiveKit dynamically adjusts stream quality based on network conditions, preventing interruptions in the call.
3. Fallback Mechanisms: When direct connections fail, TURN servers step in to relay media, ensuring the call continues uninterrupted.
Example: A Call in Action with Ports and Protocols
Below is a simplified call flow illustrating how LiveKit handles a participant joining and publishing tracks, including the ports and protocols used:
Resources
For more detailed information, check out the following LiveKit resources:
• LiveKit Official Documentation
• LiveKit Self-Hosting Guide
• LiveKit Client SDKs
• LiveKit GitHub Repository
• LiveKit Architecture Overview
These resources provide in-depth guides, tutorials, and API references for building robust applications with LiveKit.
Conclusion
LiveKit simplifies the complexities of real-time communication, providing developers with a robust framework for building a scalable and reliable systems.
Subscribe to stay tuned for my next article. In it, we will go through deploying the livekit server in a production environment on a cloud server.
Akash Gupta
Senior VoIP Engineer and AI Enthusiast
AI and VoIP Blog
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