SIP signaling and RTP transport for WaveKat voice pipelines, on a from-scratch SIP engine (no external SIP stack). Same pattern as wavekat-vad and wavekat-turn.
Warning
Early development. API will change between minor versions.
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 (digest auth + keepalive), outbound and inbound
calls (
Caller/IncomingCall), in-dialog hold/resume, DTMF (RFC 4733 + INFO fallback), and RFC 4028 session timers. - SDP: offer/answer for Opus (preferred, with in-band FEC) and G.711 (PCMU + PCMA) fallback; answers and mid-call re-offers pin the negotiated codec. Negotiation only — encode/decode stays with the consumer.
- 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.
cargo add wavekat-sipRegister an account against your SIP server:
use tokio_util::sync::CancellationToken;
use wavekat_sip::{Registrar, SipAccount, SipEndpoint, Transport};
# async fn run() -> Result<(), Box<dyn std::error::Error + Send + Sync>> {
let account = SipAccount {
display_name: "Office".into(),
username: "1001".into(),
password: "secret".into(),
domain: "sip.example.com".into(),
auth_username: None,
server: None,
port: None,
transport: Transport::Udp,
};
let cancel = CancellationToken::new();
let endpoint = SipEndpoint::new(&account, cancel.clone()).await?;
// Expires: 60s, re-register every 50s.
let registrar = Registrar::new(account, endpoint, cancel, 60, 50)?;
registrar.register().await?;
registrar.keepalive_loop().await;
# Ok(())
# }Place an outbound call and hang up:
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:[email protected]".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:
# 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(())
# }| 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 — Opus + G.711 offer/answer (negotiation only). |
rtp |
Working — header parser, receive loop, send loop. |
PSTN / SIP trunk
│
▼
wavekat-sip (in-house transport, transactions, dialogs)
│
├─ account ──── credentials + endpoint config
├─ 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 / send
│
▼
your app ──► audio device I/O, codec, recording, AI pipeline
wavekat-sip is part of WaveKat, an open-source ecosystem of Rust crates for building real-time voice pipelines. It handles SIP signaling and RTP transport, alongside sibling crates for voice activity detection, turn detection, speech-to-text, and text-to-speech.
See wavekat.com for the full project.
Licensed under Apache 2.0.
Copyright 2026 WaveKat.
rsip— SIP message types.