AIARCOASC Docs

Realtime / WebRTC

Selective forwarding unit for low-latency audio, video and data channels at the edge.

What it is

Realtime gives you an edge-hosted Selective Forwarding Unit for WebRTC: clients connect to the nearest PoP, the SFU forwards streams between them with minimal added latency. Use it for voice/video calling, multi-user games, collaborative editing.

When to use it

  • Low-latency voice/video product (calls, podcasting, livestream).
  • Multi-user collaboration with shared cursors / state.
  • Realtime telemetry from many devices to many subscribers.

Quickstart

asc edge realtime rooms create lobby --max-participants 50
room = client.edge.realtime.rooms.create('lobby', max_participants=50)
token = client.edge.realtime.rooms.token(room.id, identity='alice', publish=True, subscribe=True)
const room = await client.edge.realtime.rooms.create({ name: 'lobby', maxParticipants: 50 });
const token = await client.edge.realtime.rooms.token(room.id, { identity: 'alice' });

Limits & quotas

LimitDefaultBurstNotes
Per participant-minute$0.01Audio/video
Data channels$0.10 / GB
Max participants / room1,000Lift via sales

Pricing

See pricing. Pay-as-you-go, billed monthly via Stripe.

API surface

  • POST /v1/edge/realtime/rooms
  • POST /v1/edge/realtime/rooms/{id}/token

Required scope(s): edge:write. See Scopes.

Security

All access is authenticated and scoped. See Auth & scopes and Network controls.