v2.0.0 — Telephony Reliability, SIP, WebRTC & Asterisk

What's Changed

Telephony: Rebuilt from the Ground Up

• Unified channel architecture shared across Twilio, Vonage, Exotel, Asterisk, and SIP
• Interruptions, end-of-call signals, and transfer/hangup events handled consistently across all providers
• New `call_contexts` table persists call state — async provider callbacks resolve correctly even after call ends
• Channel UUIDs propagate end-to-end for reliable transfer and hangup operations

New: SIP Integration

Full native SIP stack with RTP handling, SDP negotiation, port allocator, and session management.

New: Asterisk / AudioSocket

Native integration with Asterisk via AudioSocket and WebSocket. Inbound and outbound call flows tested.

New: WebRTC Channel

Browser-based voice with Opus codec support and gRPC signalling, sharing the same hardened base as telephony.

Audio Pipeline: Deterministic Framing

• Exact 20 ms output frames with zero per-frame heap allocations
• Atomic interruption — `ClearOutputBuffer` drains buffers and signals output writer instantly
• Per-speaker recordings split into `assistant_recording_url` + `user_recording_url`

LLM Text Aggregator

Sentence-boundary aggregation between LLM stream and TTS — reduces first-word latency with configurable delimiters and clean context-switch flush.

Test Coverage

• 31 `BaseStreamer` unit tests
• Full telephony provider test suite (Twilio, Vonage, Exotel)
• Transformer tests for AssemblyAI, Azure, Cartesia, Deepgram, ElevenLabs, Google, Resemble, Sarvam
• LLM text aggregator: 972 lines of unit tests + 381 lines of benchmarks

Bug Fixes

• Google TTS stale response fix for outputs > 5 sentences
• AgentKit executor stability fixes
• First-token response time now tracked in LLM telemetry
• MCP tool support for agent tool invocations

Breaking Changes / Migrations

Migration	Change
`000005`	New `call_contexts` table required
`000006`	`recording_url` split into `assistant_recording_url` + `user_recording_url`