A comprehensive, pure Rust SIP/VoIP stack -- 15 crates, ~397,000 lines of Rust, 3,183 tests

Quick Start | Architecture | Examples | API Docs

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rvoip is a pure Rust SIP/VoIP stack built from the ground up for building softphones, SIP servers, and enterprise call centers. All 14 SIP methods have production handlers. Four transport protocols, eight security modules, and four audio codecs ship complete. Seven production library adapters (webrtc-rs, stun-rs) back the security and NAT traversal layers.

Quick Start

Add rvoip to your Cargo.toml:

[dependencies]
rvoip = { version = "0.1", features = ["full"] }
tokio = { version = "1.0", features = ["full"] }

SIP Server

use rvoip::session_core::prelude::*;

#[tokio::main]
async fn main() -> Result<()> {
    let session_manager = SessionManagerBuilder::new()
        .with_sip_port(5060)
        .build().await?;

    println!("SIP server running on port 5060");
    tokio::signal::ctrl_c().await?;
    Ok(())
}

Make a Call

use rvoip::client_core::{ClientConfig, ClientManager, MediaConfig};

#[tokio::main]
async fn main() -> Result<(), Box<dyn std::error::Error>> {
    let config = ClientConfig::new()
        .with_sip_addr("127.0.0.1:5060".parse()?)
        .with_media_addr("127.0.0.1:20000".parse()?)
        .with_user_agent("MyApp/1.0".to_string())
        .with_media(MediaConfig {
            preferred_codecs: vec!["PCMU".to_string(), "PCMA".to_string()],
            ..Default::default()
        });

    let client = ClientManager::new(config).await?;
    client.start().await?;

    let call_id = client.make_call(
        "sip:alice@127.0.0.1".to_string(),
        "sip:bob@example.com".to_string(),
        None
    ).await?;

    println!("Call initiated to bob@example.com");
    Ok(())
}

Call Center

use rvoip::call_engine::{prelude::*, CallCenterServerBuilder};

#[tokio::main]
async fn main() -> Result<(), Box<dyn std::error::Error>> {
    let mut config = CallCenterConfig::default();
    config.general.local_signaling_addr = "0.0.0.0:5060".parse()?;
    config.general.domain = "127.0.0.1".to_string();

    let mut server = CallCenterServerBuilder::new()
        .with_config(config)
        .with_database_path(":memory:".to_string())
        .build()
        .await?;

    server.start().await?;
    server.run().await?;
    Ok(())
}

See the examples/ directory for complete working applications.

Architecture

┌─────────────────────────────────────────────────────────────┐
│                    Application Layer                        │
│  ┌──────────────┐  ┌──────────────┐  ┌──────────────┐      │
│  │ call-engine  │  │ client-core  │  │  sip-client  │      │
│  │(Call Center) │  │ (SIP Client) │  │(Simple API)  │      │
│  └──────────────┘  └──────────────┘  └──────────────┘      │
├─────────────────────────────────────────────────────────────┤
│               Session & Coordination Layer                  │
│                   ┌─────────────────┐                       │
│                   │  session-core   │                       │
│                   │ (Session Mgmt)  │                       │
│                   └─────────────────┘                       │
├─────────────────────────────────────────────────────────────┤
│               Protocol & Processing Layer                   │
│  ┌─────────────────┐  ┌─────────────────┐                   │
│  │   dialog-core   │  │   media-core    │                   │
│  │  (SIP Dialogs   │  │ (Audio Process) │                   │
│  │  + Transactions) │  └─────────────────┘                   │
│  └─────────────────┘                                        │
├─────────────────────────────────────────────────────────────┤
│               Transport & Media Layer                       │
│  ┌─────────────────┐  ┌─────────────────┐                   │
│  │ sip-transport   │  │   rtp-core      │                   │
│  │ (SIP Transport) │  │ (RTP/SRTP)      │                   │
│  └─────────────────┘  └─────────────────┘                   │
├─────────────────────────────────────────────────────────────┤
│                    Foundation Layer                         │
│  ┌─────────────────┐  ┌──────────────┐  ┌──────────────┐   │
│  │    sip-core     │  │  codec-core  │  │  audio-core  │   │
│  │ (SIP Protocol)  │  │  (Codecs)    │  │  (Audio DSP) │   │
│  └─────────────────┘  └──────────────┘  └──────────────┘   │
├─────────────────────────────────────────────────────────────┤
│                    Support Crates                           │
│  ┌───────────────┐ ┌────────────────┐ ┌──────────────────┐ │
│  │ infra-common  │ │ registrar-core │ │ intermediary-core│ │
│  └───────────────┘ └────────────────┘ └──────────────────┘ │
│  ┌───────────────┐                    ┌──────────────────┐ │
│  │  users-core   │                    │      rvoip       │ │
│  └───────────────┘                    │    (facade)      │ │
│                                       └──────────────────┘ │
└─────────────────────────────────────────────────────────────┘

Core Crates

Crate	LOC	Tests	Description
sip-core	120,829	2,002	SIP protocol foundation -- RFC 3261 parsing, serialization, 60+ headers, SDP
rtp-core	64,931	377	RTP/RTCP/SRTP/DTLS-SRTP, ICE/STUN/TURN, ZRTP, MIKEY
dialog-core	43,864	173	SIP dialog state machine + transaction layer
session-core	42,214	82	Session lifecycle management, SIP-media coordination
media-core	34,603	245	Audio processing -- AEC, AGC, VAD, noise suppression
call-engine	28,204	6	Call center orchestration -- B2BUA, queuing, routing
client-core	19,918	27	High-level SIP client API
infra-common	9,547	26	Shared infrastructure utilities
codec-core	7,043	106	G.711, G.722, G.729A (pure Rust), Opus (feature-gated)
audio-core	6,526	56	Audio DSP and device I/O
sip-transport	6,243	41	UDP, TCP, TLS, WebSocket transport
sip-client	6,166	40	Simplified SIP client with simple-api feature
registrar-core	2,287	--	SIP registrar and contact management
intermediary-core	1,323	--	B2BUA and proxy functionality
rvoip (facade)	--	--	Re-exports all crates under rvoip::*

Total: ~397,000 lines of Rust, 3,183 tests passing

SIP Protocol Features

All 14 SIP methods have production handlers in dialog-core.

Method	Status	RFC	References
INVITE	✅ Complete	RFC 3261	107
ACK	✅ Complete	RFC 3261	24
BYE	✅ Complete	RFC 3261	20
CANCEL	✅ Complete	RFC 3261	18
REGISTER	✅ Complete	RFC 3261	10
OPTIONS	✅ Complete	RFC 3261	6
UPDATE	✅ Complete	RFC 3311	24
SUBSCRIBE	✅ Complete	RFC 6665	3
NOTIFY	✅ Complete	RFC 6665	9
MESSAGE	✅ Complete	RFC 3428	11
INFO	✅ Complete	RFC 6086	22
REFER	✅ Complete	RFC 3515	13
PRACK	✅ Complete	RFC 3262	2
PUBLISH	✅ Complete	RFC 3903	3

Security and Encryption

Module	Status	LOC	Details
SRTP/SRTCP	✅ Complete	3,218	AES-CM, HMAC-SHA1, AES-128-GCM, AES-256-GCM + webrtc-srtp 0.17.1 adapter
DTLS	✅ Complete	11,531	Full handshake and cipher activation + webrtc-dtls 0.12.0 adapter
ICE	✅ Complete	4,008	Full agent, trickle ICE, consent freshness + webrtc-ice 0.17.1 adapter
STUN	✅ Complete	2,175	Binding requests, NAT detection + stun-rs 0.1.11 adapter
TURN	✅ Complete	1,756	Relay allocation, channel binding
SCTP	✅ Complete	1,654	DTLS-SCTP data channels + webrtc-sctp 0.17.1 adapter
ZRTP	✅ Complete	1,715	DH key exchange, SAS verification
MIKEY	✅ Complete	2,357	PSK, PKE, DH (ECDH P-256) -- all three modes
Digest Auth	✅ Complete	927	MD5, SHA-256 (RFC 2617/7616)
TLS Transport	✅ Complete	769	TLS 1.2/1.3 for SIP signaling

Transport

Protocol	Status	LOC	Notes
UDP	✅ Complete	528	Primary SIP transport
TCP	✅ Complete	1,129	Reliable SIP transport
TLS	✅ Complete	769	TLS 1.2/1.3 secure signaling
WebSocket	✅ Complete	1,402	WS + WSS, client + server (RFC 7118)

NAT Traversal

Feature	Status	RFC
ICE	✅ Complete	RFC 8445 -- full agent, trickle ICE, consent freshness
STUN	✅ Complete	RFC 5389 -- binding requests, NAT detection
TURN	✅ Complete	RFC 5766 -- relay allocation, channel binding
Symmetric RTP	✅ Complete	RFC 4961

Audio Codecs

Codec	Status	LOC	Implementation
G.711 PCMU/PCMA	✅ Complete	406	Pure Rust, u-law/A-law, 8 kHz
G.722	✅ Complete	463	Pure Rust, ADPCM + QMF sub-band, 16 kHz
G.729A	✅ Complete	1,786	Pure Rust, CS-ACELP
Opus	✅ Complete	544	Feature-gated libopus binding

Production Library Adapters

rvoip writes its own protocol implementations and adapts to battle-tested ecosystem crates for production deployment:

Adapter	Target Library	LOC
ICE	webrtc-ice 0.17.1	1,056
STUN	stun-rs 0.1.11	600
RTP/RTCP	rtp/rtcp 0.17.1	573
SCTP	webrtc-sctp 0.17.1	450
DTLS	webrtc-dtls 0.12.0	377
SRTP	webrtc-srtp 0.17.1	260
Audio DSP	webrtc-audio-processing 2.0	481 (feature-gated)

Testing

4-Level Test Architecture

Level	Scope	Command
L1 Unit	Per-crate isolated tests	cargo test -p <crate>
L2 Adapter	Production library adapter roundtrips	./scripts/test_all.sh adapter
L3 Integration	Cross-crate module integration	./scripts/test_all.sh integration
L4 End-to-End	Complete call paths with audio	./scripts/test_all.sh e2e

Test Results

Crate	Passed
sip-core	2,002
rtp-core	377
media-core	245
dialog-core	173
codec-core	106
session-core	82
audio-core	56
sip-transport	41
sip-client	40
client-core	27
infra-common	26
call-engine	6
Total	3,183

Adapter roundtrip tests cover all 7 production library integrations. RFC 4475 torture tests validate SIP parser compliance.

Known Gaps

• Video codecs -- No H.264/VP8/VP9 support. rvoip is audio-only.

Roadmap

• Video codec support (H.264, VP8, VP9)
• Mobile SDKs (iOS and Android via FFI)
• Clustering and high availability
• WebRTC gateway (browser-to-SIP interop)
• REST/GraphQL management API
• REST/GraphQL management API

Contributing

Contributions welcome. Open an issue for bugs or feature requests, or submit a pull request.

License

Licensed under either of:

• Apache License, Version 2.0
• MIT License

at your option.