feat!: in-house SIP engine, full call lifecycle

README.md

@@ -7,8 +7,8 @@
 [![Crates.io](https://img.shields.io/crates/v/wavekat-sip.svg)](https://crates.io/crates/wavekat-sip)
 [![docs.rs](https://docs.rs/wavekat-sip/badge.svg)](https://docs.rs/wavekat-sip)
 
-SIP signaling and RTP transport for [WaveKat](https://wavekat.com) voice pipelines, built on
-[`rsipstack`](https://crates.io/crates/rsipstack). Same pattern as
+SIP signaling and RTP transport for [WaveKat](https://wavekat.com) voice
+pipelines, on a from-scratch SIP engine (no external SIP stack). Same pattern as
 [wavekat-vad](https://github.com/wavekat/wavekat-vad) and
 [wavekat-turn](https://github.com/wavekat/wavekat-turn).
 
@@ -20,11 +20,12 @@ SIP signaling and RTP transport for [WaveKat](https://wavekat.com) voice pipelin
 A small, focused SIP/RTP toolkit for building softphones, voice bots, and
 recording bridges in Rust. It owns the wire-level concerns —
 
-- **SIP signaling**: REGISTER (with digest auth + keepalive), INVITE (in/out),
-  BYE, dialog tracking.
-- **SDP**: minimal offer/answer for G.711 telephony audio.
-- **RTP**: header parser and a receive loop suitable for transcription /
-  recording / debug.
+- **SIP signaling**: REGISTER (digest auth + keepalive), outbound and inbound
+  calls (`Caller` / `IncomingCall`), in-dialog hold/resume, DTMF (RFC 4733 +
+  INFO fallback), and RFC 4028 session timers.
+- **SDP**: minimal offer/answer for G.711 (PCMU + PCMA) telephony audio.
+- **RTP**: header parser, a debug-friendly receive loop, and a codec-agnostic
+  send loop.
 
 — and stays out of the audio device, codec, and call-orchestration layers
 so it remains light and embeddable.
@@ -38,7 +39,6 @@ cargo add wavekat-sip
 Register an account against your SIP server:
 
 ```rust,no_run
-use std::sync::Arc;
 use tokio_util::sync::CancellationToken;
 use wavekat_sip::{Registrar, SipAccount, SipEndpoint, Transport};
 
@@ -55,8 +55,7 @@ let account = SipAccount {
 };
 
 let cancel = CancellationToken::new();
-let (endpoint, _incoming) = SipEndpoint::new(&account, cancel.clone()).await?;
-let endpoint = Arc::new(endpoint);
+let endpoint = SipEndpoint::new(&account, cancel.clone()).await?;
 
 // Expires: 60s, re-register every 50s.
 let registrar = Registrar::new(account, endpoint, cancel, 60, 50)?;
@@ -66,34 +65,67 @@ registrar.keepalive_loop().await;
 # }
 ```
 
-INVITE wrappers (`Caller`, `Callee`) land in the next release. Until then,
-drive `SipEndpoint::dialog_layer` directly.
+Place an outbound call and hang up:
+
+```rust,no_run
+use std::sync::Arc;
+use wavekat_sip::{Caller, SipAccount, SipEndpoint};
+
+# async fn run(account: SipAccount, endpoint: Arc<SipEndpoint>)
+#     -> Result<(), Box<dyn std::error::Error + Send + Sync>> {
+let caller = Caller::new(account, endpoint);
+let target: wavekat_sip::re_exports::Uri = "sip:bob@example.com".try_into()?;
+let mut call = caller.dial(target).await?;
+
+// Wire call.rtp_socket + call.remote_media to your audio / AI pipeline, then:
+call.hangup().await?;
+# Ok(())
+# }
+```
+
+Answer inbound calls from the endpoint's incoming stream:
+
+```rust,no_run
+# use std::sync::Arc;
+# use wavekat_sip::SipEndpoint;
+# async fn run(endpoint: Arc<SipEndpoint>)
+#     -> Result<(), Box<dyn std::error::Error + Send + Sync>> {
+while let Some(incoming) = endpoint.next_incoming_call().await {
+    // Inspect incoming.remote_media, then accept (or reject):
+    let _call = incoming.accept().await?;
+}
+# Ok(())
+# }
+```
 
 ## Status
 
-| Module      | State                                                |
-|-------------|------------------------------------------------------|
-| `account`   | Stable — runtime SIP account type.                   |
-| `endpoint`  | Working — shared SIP endpoint + transport.           |
-| `registrar` | Working — REGISTER + auth + keepalive + unregister.  |
-| `sdp`       | Working — minimal G.711 offer/answer.                |
-| `rtp`       | Header parser + debug receive loop. RTP send next.   |
-| `caller`    | _Planned_ — outbound INVITE wrapper.                 |
-| `callee`    | _Planned_ — inbound INVITE accept/reject helper.     |
+| Module      | State                                                  |
+|-------------|--------------------------------------------------------|
+| `account`   | Stable — runtime SIP account type.                     |
+| `endpoint`  | Working — shared SIP endpoint + transport + routing.   |
+| `registrar` | Working — REGISTER + auth + keepalive + unregister.    |
+| `resolve`   | Working — RFC 3263 (subset) SRV + A/AAAA fallback.     |
+| `caller`    | Working — outbound dial, hold/resume, DTMF, hangup.    |
+| `callee`    | Working — inbound INVITE accept/reject.                |
+| `sdp`       | Working — minimal G.711 offer/answer.                  |
+| `rtp`       | Working — header parser, receive loop, send loop.      |
 
 ## Architecture
 
 ```
 PSTN / SIP trunk
        │
        ▼
-   wavekat-sip   ──►  rsipstack (transport, transactions, dialogs)
+   wavekat-sip   (in-house transport, transactions, dialogs)
        │
        ├─ account ──── credentials + endpoint config
-       ├─ endpoint ─── UDP/TCP transport + DialogLayer
+       ├─ endpoint ─── UDP transport + transaction/dialog engine + routing
        ├─ registrar ── REGISTER / digest auth / keepalive
+       ├─ caller ───── outbound INVITE / hold / DTMF / hangup
+       ├─ callee ───── inbound INVITE accept / reject
        ├─ sdp ──────── offer/answer for telephony codecs
-       └─ rtp ──────── RTP header parse + receive
+       └─ rtp ──────── RTP header parse / receive / send
        │
        ▼
    your app  ──► audio device I/O, codec, recording, AI pipeline
@@ -122,6 +154,4 @@ Copyright 2026 WaveKat.
 
 ### Acknowledgements
 
-- [`rsipstack`](https://crates.io/crates/rsipstack) — the SIP transaction /
-  dialog engine this crate wraps.
 - [`rsip`](https://crates.io/crates/rsip) — SIP message types.

crates/wavekat-sip/Cargo.toml

@@ -13,18 +13,16 @@ categories = ["network-programming", "multimedia::audio"]
 build = "build.rs"
 
 [dependencies]
+# Only `rsip` (SIP message types) is used; the transaction/dialog/transport
+# engine is built in-house under `src/stack/` (see docs/08-own-sip-stack.md).
 rsip = "0.4"
-rsipstack = "0.4"
 tokio = { version = "1", features = ["macros", "rt", "net", "sync", "time"] }
 tokio-util = { version = "0.7", features = ["rt"] }
 tracing = "0.1"
 serde = { version = "1", features = ["derive"] }
 thiserror = "2"
 hostname = "0.4"
 # DNS SRV lookups for RFC 3263 server location (src/resolve.rs).
-# Already in our tree via rsipstack's default `srv_lookup` feature
-# (same version, same default features), so this adds no new
-# transitive deps.
 hickory-resolver = "0.25"
 
 [dev-dependencies]