Concepts on Ulai Docs

Call flows

Mon, 01 Jan 0001 00:00:00 +0000

Both legs end in the same place: a bridgedCall handed to sfugw.Run, which moves audio until one side stops. Everything before that differs.

Inbound

The gateway is the answering party. telephony.SIPRuntime invokes gateway.OnInvite for every out-of-dialog INVITE.

 carrier gateway resolver control plane
 │ INVITE ────────▶ │ │ │
 │ │ 1. Resolve ───────────▶│ │
 │ │ (host, number, IP) │ │
 │ ◀── 4xx/5xx ──── │ ◀── reject ────────────│ │
 │ │ │ │
 │ │ 2. CreateSession ──────┼────────────────▶│
 │ ◀── 200 OK ───── │ 3. Accept │
 │ ──── ACK ──────▶ │ │
 │ │ 4. PublishEvent CALL_ANSWERED ─▶ resolver│
 │ │ 5. JoinSession ────────┼────────────────▶│
 │ ◀═══ RTP ══════▶ │ ◀════════ bridge ══════╪═════ WebRTC ═══▶│

Route. (host, dialled number, source IP) → project, trunk ACL, dispatcher rule. A rejection here is a specific SIP status the carrier can act on, not a blanket 503. See routing.
Room. An X-Session-Id header on the INVITE wins if present; otherwise a session is created, carrying the dispatcher rule in its metadata so the platform can see what the room is for. The header only chooses the room — it cannot overrule admission.
Answer. 200 OK + ACK, which binds RTP.
Tell. CALL_ANSWERED is published before the WebRTC handshake, not after. That event is what summons the agent, so the dispatcher gets to work while the gateway is still joining.
Join and bridge. A seat is taken in the room, then audio flows.

The room is joined only after the call is answered. On this path “accepted” is the pickup, so no seat is taken until there is a live call to put in it — a join ticket is only valid for about 30 seconds, and a participant appearing in the room is the agent’s cue to start talking.

Routing plane

Mon, 01 Jan 0001 00:00:00 +0000

Everything the gateway knows about numbers, trunks and agents comes from ulai-sip-resolver over the platform’s Redis. The gateway holds no routing table of its own.

inbound : (host, dialled number, source IP) → project, trunk ACL, dispatcher rule
outbound : (project, trunk id) → address, transport, digest auth

Neither leg carries trunk details on the wire. Callers used to send host, port, transport and credentials on every origination, which put carrier passwords in every dialler’s logs and meant a trunk migration had to be rolled out to each of them.

Media pipeline

Mon, 01 Jan 0001 00:00:00 +0000

The audio path

The carrier speaks 8 kHz G.711. The SFU speaks 48 kHz Opus. Everything between the two happens inside the gateway:

uplink caller μ-law 8k ─decode─▶ PCM 8k ─resample─▶ PCM 48k ─encode─▶ Opus ─▶ room
downlink room Opus 48k ─decode─▶ PCM 48k ─resample─▶ PCM 8k ─mix─▶ μ-law ─▶ caller

Both directions run on a 20 ms cadence: 160 μ-law bytes on the wire, 960 samples per Opus frame. Opus is encoded at 24 kbit/s with complexity 5 — wideband speech without maxing out a CPU that may be carrying many concurrent calls.