[Roadmap 18.7] SIP outbound dial provider (alternative to Twilio)

[furukama]

SIP outbound dial provider for the R18 (#561) voice channel — alternative to the Twilio Programmable Voice impl in R18.1 (#532), for businesses that already have SIP infrastructure (PBX, SIP trunks, cloud SIP providers like Sonetel / Voxbone / Sonus).

Why this is a separate provider

• Business reality: many enterprises already pay for SIP trunks and PBX systems. Sending outbound voice through Twilio means duplicate capacity + duplicate billing. SIP integration uses what's already there.
• Per-minute economics: SIP trunks often beat Twilio's rates for high-volume outbound (sales-call agents, notification fleets, debt-collection robocalls).
• Compliance / data residency: some EU customers prefer SIP through a local provider over US-anchored Twilio infrastructure.
• Different lifecycle and auth (registrar URL, SIP user/password or client certs vs Twilio API tokens) — peer implementation, not a refinement of R18.1.

Acceptance criteria

• Implements the same outbound-dial interface as the Twilio provider (R18.1 / #532). Skill code calling dial(agent_id, e164_number) should not need to know which provider answers.
• Auth via SIP credentials in the encrypted runtime secret store (registrar URL, SIP user/password OR client cert).
• dial() initiates a SIP INVITE through the registered trunk to the given E.164 number; returns a call_id.
• Cost recorded in UsageTotals (per-minute or per-call basis depending on operator's SIP contract).
• TTS pipeline reuse (R18.2 / #533) and STT for callee responses (R18.3 / #534) work the same way through SIP audio streams (RTP).
• Audit-log entries (R18.5 / #536) — voice.call.start, voice.call.end — emitted with SIP-specific call IDs (Call-ID header).
• Unit tests using a SIP mock (e.g. sipp or a mocked SIP client library).
• Operator docs covering SIP trunk setup, common providers (FreeSWITCH / Asterisk / cloud SIP), TLS-SRTP for encrypted calls.

Depends on

• R18.1 (#532) Twilio outbound dial — primarily to define the dial() interface contract
• R18.2 (#533) TTS pipeline reuse, R18.3 (#534) STT for callee responses

Out of scope (future tickets)

• Inbound SIP (agent receives calls from a SIP trunk) — different scope, would be tracked under R18 or as a new R-item once outbound is shipped.
• WebRTC as a transport — different protocol family; can be a peer to SIP later.

Priority

P2 — same priority as the rest of R18 (channel expansion, not core wedge), but enables B2B-grade voice deployments that Twilio alone doesn't cover well.

[furukama]

EU SIP provider scope. When this lands, the "common providers" docs should ship explicit configuration templates for the three primary German B2B SIP options: Telekom Magenta Business SIP, Vodafone OneNet, and 1&1 Versatel. Each template documents trunk URL, registration model (registrar vs static IP), TLS-SRTP setup, and common gotchas. Operator-config recipes — most of the work is documentation + reference YAML, not new code. Strategic for German B2B prospecting.