Media pipeline

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.

The downlink mixes: a room can hold several publishers, and the caller gets all of them summed into one stream. Resampling is go-audio-resampler (pure Go, SIMD-enabled); μ-law and A-law conversion is zaf/g711. The only native code in the binary is libopus.

Codec negotiation

The SDP profile is deliberately narrow — the audio pipeline is μ-law 8 kHz, so accepting anything else would be a lie:

• PCMU (μ-law, PT 0) or PCMA (A-law, PT 8). When the carrier negotiates A-law, the RTP loops transcode A-law↔μ-law at the wire boundary so everything above stays μ-law.
• telephone-event (RFC 4733 DTMF) on whatever payload type the carrier assigns.
• No Opus on the SIP side, no video, no multiple m= lines.

SRTP

SDES only, with the two common profiles: AES_CM_128_HMAC_SHA1_80 (Twilio’s default) and _32. Whether it is offered is decided by the trunk’s transport — see routing.

Inbound RTP: reorder, dedupe, conceal

A conventional jitter buffer imposes a fixed delay on every packet. For a voice agent that delay is charged to time-to-first-token on every turn, including the overwhelming majority where the network was perfectly ordered. So the inbound buffer is not a fixed-delay design:

Concealment repeats the previous frame, attenuated, decaying to silence over a few frames. Repeating preserves the spectral envelope so an ASR hears a brief smear rather than the click-and-jump digital silence produces; decaying stops a lost burst becoming an audible buzz.

The cost is paid only by calls that actually have a disordered network.

Outbound RTP: a small playout cushion

The provider’s writer and the RTP writer are two independent 20 ms tickers. Without a cushion, ordinary scheduler jitter forces silence into the middle of speech. The playout buffer builds 60 ms (3 frames) before starting and caps added latency at 160 ms (8 frames), dropping the oldest beyond that.

Symmetric RTP and the source gate

Carriers behind NAT routinely send RTP from a port they never advertised in SDP, so the first well-formed packet latches the peer address and the writer re-targets to it. Everything after that is checked against the latch.

That check matters because the RTP port range is a few hundred even ports cycled round-robin: a call that ends while its carrier is still streaming leaves packets in flight that land on whichever call binds that port next. Before the gate existed, they were decoded and mixed into a live conversation as a second voice.

A genuine media re-anchor — a B2BUA leg swap, an SBC failover — is admitted only after the new source proves persistence (5 packets over at least 200 ms). A re-INVITE can pre-authorise an address, but the previous peer stays valid until the new one actually speaks: a re-INVITE is an intention, and a peer that never follows through must not be able to mute a working call.

Re-INVITE, hold, and session timers

sipgo’s OnInvite fires for every INVITE, including in-dialog ones. Handing those to the application handler treats a mid-call re-INVITE — a session-timer refresh, a hold, a media re-anchor, an SBC failover — as a brand new call: a second agent, a second billing row, a 180 Ringing inside an established dialog, a 200 OK with a new To tag (a protocol violation), and an answer advertising a new RTP port, so the carrier moves media to a socket nobody reads. Dead air for the rest of the call.

In-dialog INVITEs are therefore handled separately, and answered as a re-statement rather than a negotiation: same To tag, same RTP port, same codec, session id unchanged with its version bumped, and the direction attribute mirrored so hold is acknowledged rather than contradicted. The only thing that may legitimately change is where the peer wants media — and that is followed.

A re-INVITE that tries to switch G.711 flavour mid-call is refused rather than silently answered with a lie: the provider’s transcode setting is fixed when it is constructed.

Session timers (timer) are supported; an INVITE that Requires an extension the gateway does not support is rejected rather than answered.

DTMF

RFC 4733 telephone-event packets are decoded and surfaced as events. The gateway logs them (DTMF: 5) and does nothing else with them — it has no IVR of its own. An agent in the room that wants digits should consume the session feed, not expect the gateway to act.

Ending a call without a BYE

A hangup’s BYE can be lost. A TLS trunk calling back a UDP-only listener never reaches the gateway at all, and without a backstop such a call — and its room — stays up until the process exits.

SIP_RTP_TIMEOUT_SECONDS (default 30) ends a call whose inbound audio has stopped for that long, as if the far end had hung up. It is paused while the call is on hold, so a legitimately silent leg is not cut off. Set 0 to disable.

The other end-of-call signals: