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Compute
WebRTC media server / TURN
Relays and routes live audio/video: TURN servers get calls through firewalls and an SFU forwards streams in group calls.
When to use
- Video consultations, voice calls, live classes
Pitfalls
- Bandwidth-heavy and latency-sensitive — deploy close to users
- Recording needs a separate pipeline
Options by provider
The first option is what diagrams show for that provider.
Amazon Web Services
- Amazon Chime SDKManaged
- Kinesis Video Streams WebRTCManaged
Google Cloud
- LiveKit / coturn on GKE — no dedicated managed serviceSelf-hosted
Microsoft Azure
- Azure Communication ServicesManaged
Open source / self-hosted
- LiveKitSelf-hosted
- coturn — STUN/TURNSelf-hosted
- mediasoupSelf-hosted
- JanusSelf-hosted
Third-party (works with any cloud)
- Twilio VideoManaged
- AgoraManaged
- DailyManaged
Related concepts
Used in
- Video calling (WebRTC)
- STUN / TURN relays — NAT discovery and relays for blocked connections
- SFU media servers (Go) — Forward each stream to the other participants
- Practo / teleconsultation
- TURN / media servers — Relays calls on restrictive networks
- AstroTalk
- Calls (WebRTC / telephony) — Voice and video sessions