Crate rsiprtp 

rsiprtp - SIP/RTP stack for Rust

An audio-focused SIP user-agent (UA) stack designed for:

• Voicemail applications
• AI agent call bridges with mixing

rsiprtp targets traditional VoIP / SIP-trunking use cases. It is not a WebRTC stack: there is no DTLS-SRTP handshake (SDES per RFC 4568), no video, and no SIP-over-WebSocket transport. See the README for the full scope.

Carrier-interop signalling features are supported: PRACK / 100rel reliable provisional responses (RFC 3262, no offer/answer body in PRACK; reliable provisional acks only), the UPDATE method (RFC 3311), and session timers (RFC 4028) — both the refresher path (UPDATE / re-INVITE refresh) and the non-refresher path (BYE on peer silence) drive from the CallManager::tick / next_deadline hooks.

Auto-detection of Allow: lacking UPDATE on the 200 OK is not implemented; the app calls session::CallManager::note_update_unsupported on observing a 405 / 501 to UPDATE, or after parsing the 200 OK’s Allow header itself.

Quick Start

use rsiprtp::prelude::*;

#[tokio::main]
async fn main() -> Result<(), Error> {
    // Create a call manager
    let config = ManagerConfig::default();
    let mut manager = CallManager::new(config);

    // Create an outbound call
    let call_id = manager.create_call("sip:bob@example.com".to_string());

    // ... handle call events ...

    Ok(())
}

Architecture

The stack is organized into modules:

• core: Common types, errors, configuration
• sip: SIP message parsing and building (in-tree parser)
• transaction: RFC 3261 transaction state machines (Sans-IO)
• dialog: Dialog management for INVITE sessions
• transport: UDP/TCP/TLS network transport
• sdp: SDP parsing and offer/answer negotiation
• rtp: RTP packet handling
• srtp: SRTP encryption with SDES key exchange (RFC 4568)
• ice: ICE/STUN/TURN for NAT traversal (host + server-reflexive candidates; TURN relay candidates, trickle, ICE restart, IPv6 dual-stack interop, symmetric-NAT prflx, and RFC 7675 consent freshness are not yet supported). Drive it via session::IceSession alongside session::CallManager.
• media: Audio codecs and jitter buffer
• session: High-level call management

Modules

core

Core types, errors, and configuration for the rsiprtp SIP/RTP stack.

dialog

SIP dialog management.

ice

STUN/ICE implementation for NAT traversal.

media

Audio processing: codecs, jitter buffer, mixing, and file I/O.

prelude

Prelude for convenient imports.

rtp

RTP/RTCP packet handling per RFC 3550.

sdp

SDP (Session Description Protocol) handling.

session

High-level call and session management.

sip

SIP message parsing and building for rsiprtp.

srtp

SRTP (Secure Real-time Transport Protocol) implementation.

transaction

RFC 3261 SIP transaction layer implementation.

transport

Network transport layer for SIP signaling.