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Transport architecture

Orbit’s real-time media rides two planes, split by where the media originates, not by a per-call flag you set:
  • Orbit Media (SFU) carries WebRTC media — video rooms and the browser softphone.
  • Jambonz SBC + RTPengine carries PSTN/SIP carrier media — inbound DIDs and outbound calls.
A single call can span both planes (a browser softphone dialling a PSTN number rides Orbit Media for the browser leg and is bridged through Jambonz to the carrier). There is no transport request field and no per-call SFU-vs-bridge decision the API makes on your behalf — the plane follows directly from the endpoint and the leg type.

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TL;DR

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Orbit Media (the SFU)

Orbit Media is Orbit’s Selective Forwarding Unit. It is forked from LiveKit OSS under Apache-2 — see Attribution. It handles everything that needs server-side fan-out or mixing of WebRTC media: simulcast, per-subscriber bitrate adaptation, codec negotiation, DTLS/SRTP, and server-side recording via egress. Clients never talk to the SFU with a long-lived key. The API mints a short-lived join token scoped to a single room:
  • Video — POST /api/v1/video/rooms and /api/v1/video/rooms-scheduled create a room and mint a participant token against Orbit Media. There is no non-SFU mode for video.
  • Voice (browser softphone) — POST /api/v1/voice/softphone/token mints an Orbit Media token for a deterministic per-user room (softphone:<tenant>:<user>), so reconnects are idempotent. See Softphone token.
Recording for SFU sessions is produced by Orbit Media’s egress pipeline.

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Jambonz SBC + RTPengine (carrier legs)

PSTN and SIP carrier media never touches the SFU. It terminates on the Jambonz session border controller, with RTPengine bridging RTP/SRTP between the carrier and Orbit:
  • Inbound — Telnyx and DIDWW DIDs route through Jambonz, which applies the tenant’s number-handler routing.
  • Outbound (MT) — POST /api/v1/voice/calls initiates a call through Jambonz, which terminates it on the Devotel wholesale softswitch. Outbound voice exits only via Devotel — Orbit never wires a carrier’s call-control API for outbound termination.
In-call voice recording on this plane is SIPREC (POST /api/v1/voice/calls/:id/recording/start), not SFU egress.

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Browser softphone

The browser softphone has two client transports, chosen by the SDK in use — not by a request flag:
  • Orbit Media (WebRTC) — the dashboard softphone calls POST /api/v1/voice/softphone/token and joins an Orbit Media room.
  • SIP over WSS — the @devotel/orbit-web-sdk Softphone calls POST /api/v1/voice/softphone/register to obtain a short-lived SIP credential and REGISTERs against Orbit’s WebRTC-to-SIP gateway over a secure WebSocket. The SIP password itself is HMAC-derived and never stored, but issuance and revocation metadata (issue/expiry times, the owning org, and a revoked flag) is persisted server-side so a tenant can revoke a credential before it expires. Call DELETE /api/v1/voice/softphone/credentials/:username to revoke one — the gateway rejects the HMAC on its next validation even while it’s still within its lifetime.
Either way the carrier leg, when one exists, is bridged through Jambonz as described above.

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What the API does not expose

To keep this page honest against the code, a few things that earlier drafts implied but the API does not do:
  • There is no transport field on the POST /api/v1/voice/calls body and no INVALID_TRANSPORT error code. The plane is implied by the endpoint and leg type.
  • There is no client-facing “decision tree” service that ranks SFU vs. WebSocket bridge per call, and no per-call latency-budget selection.

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Cross-references