LiveKit vs Janus vs Mediasoup: Choosing a WebRTC Media Server

Selecting a WebRTC media server is a decision that affects your system architecture for years. The wrong choice means rewriting your signaling layer, renegotiating your infrastructure costs, or hitting scaling walls at 500 users instead of 50,000. LiveKit, Janus, and Mediasoup each represent a different point in the design space — developer experience versus flexibility versus raw performance. This post compares them on the dimensions that matter in production.

Architecture Fundamentals

All three are Selective Forwarding Units (SFUs). An SFU receives each participant's media streams and selectively forwards them to other participants without transcoding. This is the right architecture for multi-party conferencing: it scales to hundreds of participants without the CPU overhead of an MCU (mixing server).

LiveKit

LiveKit is a Go-based SFU with opinionated signaling built on Protocol Buffers over WebSocket. It ships as a single binary with built-in room management, participant state, data channels, and egress/ingress pipelines. LiveKit Cloud is the managed offering; self-hosted LiveKit is Apache 2.0 licensed.

Architecture: stateless SFU nodes behind a Redis-backed cluster coordinator. Each node handles media independently; Redis propagates room state changes across nodes. Horizontal scaling requires only adding nodes — no configuration changes.

Janus

Janus (by Meetecho) is a C-based general-purpose WebRTC gateway with a plugin architecture. It does not implement SFU semantics natively — instead, it exposes a low-level API where each plugin handles a specific use case (VideoRoom for conferencing, SIPGateway for SIP bridging, AudioBridge for audio mixing). Signaling is flexible: JSON over WebSocket, HTTP, or MQTT.

Architecture: monolithic single-process model. Scaling requires multiple Janus instances behind a load balancer, with application-level logic to route participants in the same room to the same instance.

Mediasoup

Mediasoup is a Node.js library (with a C++ worker process) that gives you raw WebRTC transport primitives — Producers, Consumers, Routers, Transports — without any signaling or room management. You build everything above the transport layer. It is the most flexible and the most work.

Architecture: single-host by default. Mediasoup's Router runs on one Worker process (one CPU core). Multi-core requires spawning multiple Workers and routing participants across them — your application code manages which Worker handles which participant.

Performance Comparison

Benchmark environment: 4-core / 16 GB VM, VP8 video at 1 Mbps, 720p 30fps, all participants sending video.

Mediasoup has the best raw performance because the C++ Worker does minimal work beyond forwarding — no room management, no signaling overhead, no state synchronization. LiveKit is close and adds significantly more operational value. Janus pays a C plugin overhead and is bottlenecked by its single-process architecture at high concurrency.

Signaling Complexity

This is the starker difference than performance.

LiveKit Signaling

LiveKit handles signaling. You call a room API and the SDK manages ICE, DTLS, offer/answer negotiation, and track subscription automatically.

// LiveKit — 20 lines to publish and subscribe
import { Room, RoomEvent, Track } from 'livekit-client';

const room = new Room();
await room.connect('wss://your-livekit-server.com', token);

// Publish camera
const localTrack = await createLocalVideoTrack();
await room.localParticipant.publishTrack(localTrack);

// Subscribe to remote tracks automatically
room.on(RoomEvent.TrackSubscribed, (track, publication, participant) => {
  const element = track.attach();
  document.body.appendChild(element);
});

Mediasoup Signaling

Mediasoup gives you transports. Your signaling server (WebSocket, SIP, whatever you want) is entirely your responsibility.

// Mediasoup server-side — create transports and handle signaling manually
const worker = await mediasoup.createWorker();
const router = await worker.createRouter({ mediaCodecs });

// For each participant joining:
const transport = await router.createWebRtcTransport({
  listenIps: [{ ip: '0.0.0.0', announcedIp: '203.0.113.1' }],
  enableUdp: true,
  enableTcp: true,
  preferUdp: true,
});

// Send transport parameters to client via YOUR signaling channel
socket.emit('transport-created', {
  id: transport.id,
  iceParameters: transport.iceParameters,
  iceCandidates: transport.iceCandidates,
  dtlsParameters: transport.dtlsParameters,
});

// Handle connect from client
socket.on('transport-connect', async ({ dtlsParameters }) => {
  await transport.connect({ dtlsParameters });
});

// Handle produce from client
socket.on('produce', async ({ kind, rtpParameters }, callback) => {
  const producer = await transport.produce({ kind, rtpParameters });
  callback({ id: producer.id });
  
  // Route to consumers (other participants) — your logic
  broadcastNewProducer(producer.id, router);
});

Mediasoup requires 500–800 lines of server-side signaling code for a basic conferencing app. LiveKit requires roughly 50. Janus sits in between — its REST API handles room creation and participant management, but you still write the WebRTC negotiation logic in your client.

Multi-Server Clustering

LiveKit's clustering story is its strongest differentiator against Janus and Mediasoup. Deploy 5 LiveKit nodes behind a load balancer, point them at the same Redis, and room participants automatically end up on the same node via the cluster coordinator. Janus and Mediasoup require your application to track which instance a room lives on and route participants accordingly — real engineering work that LiveKit gives you for free.

SDK and Ecosystem

LiveKit invests heavily in client SDKs. For mobile applications, LiveKit is the clear choice — building native iOS and Android WebRTC clients without an SDK is weeks of work.

Use Case Decision Matrix

Choose LiveKit when:

• You need mobile SDK support (iOS, Android, React Native)
• You want clustering and egress without custom infrastructure code
• Your team is building fast and wants managed or self-hosted with minimal ops burden
• You need simulcast, dynacast, and adaptive bitrate out of the box

Choose Janus when:

• You need SIP/WebRTC bridging (Janus SIPGateway plugin)
• You have specialized protocol requirements (RTMP, HLS ingest)
• Your team has C expertise and wants to write custom Janus plugins
• You're extending an existing Janus deployment

Choose Mediasoup when:

• Maximum performance per core is the constraint (streaming at scale, CDN POPs)
• You need total control over signaling and have the engineering capacity to build it
• You're integrating into an existing Node.js backend with specific signaling semantics
• You're building specialized applications (low-latency auction, live betting, real-time gaming) where SFU primitives map directly to your domain model

Infrastructure Cost Estimate

At 10,000 daily active users, 30-minute average session, 5 participants per room average:

The self-hosted Mediasoup cost advantage rarely outweighs the engineering investment unless you're operating at a scale where those infrastructure savings compound significantly (100,000+ DAU).