0370347642
Record the SIP edge decision and align the docs: - docs/adr/0001-sip-strategy.md: layered strategy (own Rust parser, rent the interop tail via a Kamailio + rtpengine SBC, grow native core behind the shield); pjproject FFI explicitly rejected for breaking the memory-safety thesis at the most exposed seam. - PORT_PLAN §1 + open decisions: SIP row updated to the decided strategy. - ARCHITECTURE: "biggest technical risk" now points at ADR-0001. Co-Authored-By: Claude Opus 4.8 (1M context) <noreply@anthropic.com> Claude-Session: https://claude.ai/code/session_01C2bfD7MkqEdfnMXxXBu456
57 lines
2.8 KiB
Markdown
57 lines
2.8 KiB
Markdown
# Rutster Architecture
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## The reframe
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Asterisk's power was: *one process, load any `.so`, wire anything to anything in the
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dialplan.* That composability is the thing to match — but it does **not** require a
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1.2M-LOC monolith. Rutster delivers the same "build anything" through a different
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substrate:
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- a **small hardened core** (media + signaling glue + call model),
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- a **WASM plugin runtime** for safe, third-party-extensible logic,
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- **declarative routing** as data for the common path,
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- a **programmable API** (REST/gRPC + event stream) modeled on Asterisk's ARI.
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More extensible than Asterisk, because extensions are safe to run from people you
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don't fully trust.
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## Three planes
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### Control plane (stateless-ish, horizontally scalable)
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The ARI-style resource API (channels / bridges / endpoints / recordings / playbacks)
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over REST + gRPC + a WebSocket/SSE event stream. Registrar, routing, auth. This is
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where "the dialplan" disappears — replaced by declarative routing + external services
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reacting to call events (the Twilio / ARI-Stasis model). Asterisk's
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`rest-api/api-docs/*.json` is a reusable spec for the resource model.
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### Media plane (stateful, latency-pinned, scaled separately)
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RTP/SRTP termination, mixing/bridging (softmix), transcoding, record/playback. A
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controllable media node driven over gRPC by the control plane. Built on the Rust
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WebRTC media ecosystem (`str0m` sans-IO design, `webrtc-rs`). **The media datapath
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stays tight** — do not over-decompose it across service hops; latency and failure
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modes compound.
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### App plane (your services + plugins, outside the core)
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IVR, queues, voicemail, dialers, custom routing — driven via the API, deployed
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independently. WASM plugins for in-call logic that needs to run close to the core;
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microservices for stateful/business/billing logic.
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## Cross-cutting
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- **Event bus** (NATS / Redis Streams / Kafka) replaces Asterisk's internal Stasis bus
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for cross-service events; a lightweight in-core dispatcher handles intra-core.
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- **State store** replaces `astdb` + realtime/sorcery.
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- **Security is load-bearing, not a row:** memory-safe fuzzed parsers, TLS/SRTP
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mandatory, deny-by-default routing + toll-fraud engine, mTLS gRPC admin (no AMI),
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WASM tenant isolation, SBOM + KMS/Vault for secrets.
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- **Observability:** OpenTelemetry traces that follow a single call across
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signaling → media → app services.
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## Biggest technical risk
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The **SIP stack** — **decided in [ADR-0001](adr/0001-sip-strategy.md)**: own the Rust
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parser from day one (the security thesis depends on it), front the public edge with a
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proven **Kamailio + rtpengine** SBC to absorb the interop tail, and grow the native Rust
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transaction/dialog core behind that shield. No pjproject FFI. Everything else builds on
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the existing Rust media ecosystem.
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