Janus vs LiveKit vs mediasoup — Which WebRTC Server Should You Choose?

Quick links (your site routing format)

• How to Add a Public Website Button That Connects to Your AI Voice Bot
• ICE vs STUN vs TURN — Complete WebRTC Networking Guide
• Janus WebRTC Gateway Installation on Ubuntu (Production Guide)
• How to Connect Janus to Asterisk Extension 7000 (SIP Plugin + ARI)

This page is the “why choose which server” comparison. The links above are “how to implement”.

The user’s world

Browser (Chrome/Safari/Edge) connects with WebRTC.

• No SIP credentials
• Works behind home Wi-Fi and mobile networks
• Requires ICE + (often) TURN to succeed

Your telecom world

PBX (Asterisk / FreeSWITCH) routes calls, records, queues, etc.

• SIP signaling for call setup
• RTP audio for media
• Often deployed on a server with strict firewall rules

Pick Janus if:

• You already have Asterisk/FreeSWITCH or will use one.
• You need a WebRTC ⇄ SIP bridge for a click-to-call button.
• You want a “telecom gateway” style product: plugins, stable, predictable.
• You value clear integration boundaries: browser ↔ gateway ↔ PBX.

Janus is widely used as a WebRTC gateway, and its SIP plugin is explicitly meant to register/call against SIP servers like Asterisk. :contentReference[oaicite:0]{index=0}

Pick LiveKit if:

• Your product is primarily WebRTC (rooms, sessions, “SFU-first”).
• You want a modern developer experience and built-in real-time features.
• You expect high concurrency and want a scaling story built around SFU.

LiveKit is SFU-based and documents self-hosted SFU architectures. :contentReference[oaicite:1]{index=1}

Pick mediasoup if:

• You want to build a custom media server product and own the internals.
• You have strong WebRTC engineering and can invest in long-term maintenance.
• You need maximum flexibility, even if it costs engineering time.

mediasoup is often used as an SFU building block (library-style), not a turnkey product.

When NOT to pick Janus / LiveKit / mediasoup

• If your goal is only PSTN calling without web browsers: you may only need SIP trunks + PBX + a bot bridge (no WebRTC server).
• If you need “Zoom-like meetings” with huge feature surface: you might use LiveKit + a separate telephony bridge rather than Janus.
• If your team is small and time is limited: avoid building SFU internals from scratch (mediasoup-heavy approach).

Layman explanation (no jargon)

• STUN is like asking a public “mirror” website: “What is my public address?”
• TURN is like renting a public “relay phone line” when direct connection fails.
• ICE is the negotiation process that tries all possible paths and picks one that works.

If your audience includes mobile users, corporate Wi-Fi, or strict firewalls, TURN becomes “not optional” for reliability.

Technical explanation (still readable)

• ICE gathers candidates: host (local), srflx (STUN), relay (TURN).
• The peers test connectivity pairs and select a working route.
• Without TURN, symmetric NAT and strict enterprise networks commonly break calls.

Full deep-dive: ICE vs STUN vs TURN

Janus (WebRTC Gateway)

Janus is a gateway with plugins. In “click-to-call + SIP”, you typically use the SIP plugin to register to a SIP server (Asterisk/FreeSWITCH) and bridge a browser WebRTC session into a SIP call. :contentReference[oaicite:2]{index=2}

• Feels like a “router” between protocols: WebRTC ↔ SIP, WebRTC ↔ streaming, etc.
• Often simpler for PBX integration than SFU-centric stacks.
• Great when your core call logic lives in SIP/PBX.

LiveKit (SFU-first real-time platform)

LiveKit is an SFU-based real-time platform. SFU means it forwards media streams efficiently without mixing/decoding in the same way as an MCU. LiveKit documents SFU architectures (single-home and mesh). :contentReference[oaicite:3]{index=3}

• Excellent for “rooms”, multi-party sessions, and built-in real-time features.
• Telephony/PBX integration typically requires additional bridging logic.
• When WebRTC is the product, LiveKit is a strong default.

mediasoup (SFU building block)

mediasoup is commonly used as a lower-level SFU component that you build around. You get flexibility, but you also own a lot of plumbing: signaling, auth, scaling, operational tooling, and telephony bridging.

• Great when you need a customized media pipeline and you can invest engineering.
• Not the fastest path if you mainly need “WebRTC to SIP gateway”.

Where Asterisk/FreeSWITCH fit

Asterisk/FreeSWITCH are SIP/PBX systems. They are excellent at: IVR, queues, ring groups, transfers, recording, trunking, compliance rules, and all the “telecom business logic”.

Your WebRTC server choice is mostly about: “How do browsers enter this SIP world reliably and safely?”

What the PBX is great at

• Inbound routing logic (DIDs, departments, time conditions)
• IVR menus
• Queues & ring groups
• Recording and compliance prompts
• Agent takeover and transfers
• PSTN integration via SIP trunks / gateways

For AI voice, PBX gives you business-safe call flows. The WebRTC layer is just the “browser entry”.

Why a gateway (Janus) usually fits PBX teams

• You can make the browser appear as a SIP endpoint (via Janus SIP plugin), so the PBX keeps its mental model: “it’s just another endpoint/call leg.” :contentReference[oaicite:4]{index=4}
• You keep WebRTC-specific complexity (DTLS/SRTP, ICE candidates, TURN) outside the PBX.
• You can scale the gateway separately from the PBX media servers.

Layman examples

• Home Wi-Fi: usually works with STUN (but not always).
• Corporate Wi-Fi: often blocks UDP; TURN/TCP or TURN/TLS is needed.
• Mobile networks: NAT is aggressive; TURN becomes important.
• International users: latency spikes if TURN is too far away.

If your business depends on “every lead can call”, TURN is part of your product.

Technical truths

• ICE tries direct paths first, but many networks require relay.
• TURN means your server relays media — plan bandwidth cost.
• Deploy TURN close to users (regions) if you need low latency.
• Use safe firewall rules and monitor TURN allocations.

Read: ICE vs STUN vs TURN

Scaling with Janus (gateway model)

• Scale Janus horizontally behind a load balancer.
• Scale TURN horizontally and regionally.
• PBX scaling depends on your call concurrency and architecture.
• For “AI bot calls”: plan where the bot media processing runs (close to PBX / close to Janus / close to TURN).

A gateway model is often easier to reason about when SIP is central.

Scaling with LiveKit/mediasoup (SFU model)

• SFUs are designed for forwarding streams efficiently.
• Good for multi-party sessions or many listeners.
• Telephony bridging can become the “special edge” you must scale carefully.
• When the bot is the “other participant”, SFU benefits can be smaller unless you have multi-party or fan-out use cases.

LiveKit documents SFU scaling architectures. :contentReference[oaicite:5]{index=5}

Minimum safe practices

• Use time-limited TURN credentials (dynamic auth).
• Rate limit session creation per IP / per token.
• Put WebRTC gateway behind a WAF where appropriate.
• Keep SIP servers (Asterisk/FreeSWITCH) not directly exposed to the public internet.
• Separate “public edge” (Janus/TURN) from “PBX core” with firewall rules.

Practical compliance notes (non-legal advice)

• Disclose recording if required by your region.
• Store recordings securely with access logs.
• Mask or tokenize sensitive data if AI sees transcripts.
• Keep audit logs for who accessed what.

Mistake #1: No TURN (or TURN only UDP)

Works for the developer at home, fails for customers. Fix: deploy TURN with UDP + TCP + TLS 443, and use time-limited credentials.

Mistake #2: Putting PBX on the public internet

PBX exposure leads to scanning and brute force. Fix: keep PBX private; expose only gateway/TURN edge with strict rules.

Mistake #3: Mixing “WebRTC media server” and “PBX routing” responsibilities

Teams try to recreate IVR/queues inside WebRTC server code. Fix: keep PBX for telecom logic; keep gateway/SFU for browser connectivity.

Mistake #4: No observability

Without metrics, you can’t tell if failures are TURN, ICE, gateway, or SIP. Fix: track allocation failures, ICE selected candidate type (host/srflx/relay), call setup timing.