Crate saorsa_webrtc_core 

Saorsa WebRTC - WebRTC implementation over ant-quic transport

This library provides a WebRTC implementation that uses ant-quic as the underlying transport layer instead of traditional ICE/STUN/TURN. It features:

• Native QUIC Transport: Uses ant-quic for reliable, encrypted connections
• DHT-based Signaling: Distributed signaling without centralized servers
• Post-Quantum Cryptography: Built-in PQC support via ant-quic
• NAT Traversal: Automatic hole punching and relay fallback
• High Performance: Low-latency media streaming with QoS

Examples

use saorsa_webrtc_core::{WebRtcService, MediaConstraints, SignalingHandler, AntQuicTransport, TransportConfig, PeerIdentityString};
use std::sync::Arc;

// Create signaling transport
let transport = Arc::new(AntQuicTransport::new(TransportConfig::default()));
let signaling = Arc::new(SignalingHandler::new(transport));

// Create WebRTC service
let service = WebRtcService::<PeerIdentityString, AntQuicTransport>::new(
    signaling,
    Default::default()
).await?;

// Start service
service.start().await?;

// Initiate a video call
let call_id = service.initiate_call(
    PeerIdentityString::new("eve-frank-grace-henry"),
    MediaConstraints::video_call()
).await?;

Re-exports

pub use call::CallManager;

pub use call::CallManagerConfig;

pub use identity::PeerIdentity;

pub use identity::PeerIdentityString;

pub use link_transport::LinkTransport;

pub use link_transport::LinkTransportError;

pub use link_transport::PeerConnection;

pub use link_transport::StreamType as LinkStreamType;

pub use media::AudioDevice;

pub use media::AudioTrack;

pub use media::MediaEvent;

pub use media::MediaStream;

pub use media::MediaStreamManager;

pub use media::VideoDevice;

pub use media::VideoTrack;

pub use protocol_handler::WebRtcHandlerConfig;

pub use protocol_handler::WebRtcHandlerError;

pub use protocol_handler::WebRtcIncoming;

pub use protocol_handler::WebRtcProtocolHandler;

pub use protocol_handler::WebRtcProtocolHandlerBuilder;

pub use quic_bridge::RtpPacket;

pub use quic_bridge::StreamConfig;

pub use quic_bridge::StreamType;

pub use quic_bridge::WebRtcQuicBridge;

pub use quic_media_transport::MediaTransportError;

pub use quic_media_transport::MediaTransportState;

pub use quic_media_transport::QuicMediaTransport;

pub use quic_media_transport::StreamHandle;

pub use quic_media_transport::StreamPriority;

pub use quic_media_transport::TransportStats;

pub use service::WebRtcConfig;

pub use service::WebRtcEvent;

pub use service::WebRtcService;

pub use service::WebRtcServiceBuilder;

pub use signaling::SignalingHandler;

pub use signaling::SignalingMessage as SignalingMessageType;

pub use signaling::SignalingTransport;

pub use transport::AntQuicTransport;

pub use transport::TransportConfig;

pub use types::*;

Modules

call

Call management and state (requires legacy-webrtc feature) Call management for WebRTC

identity

Peer identity abstraction Peer identity abstraction

link_transport

Link transport abstraction layer Link transport abstraction layer

media

Media stream management (requires legacy-webrtc feature) Media stream management for WebRTC

prelude

Prelude module for convenient imports

protocol_handler

Protocol handler for SharedTransport integration WebRTC protocol handler for SharedTransport integration

quic_bridge

Bridge between WebRTC and QUIC WebRTC to QUIC bridge

quic_media_transport

QUIC-based media transport for RTP/RTCP over QUIC streams QUIC Media Transport for WebRTC over ant-quic

quic_streams

QUIC media stream management with QoS QUIC media stream management with QoS

service

WebRTC service and configuration (requires legacy-webrtc feature) WebRTC service orchestration

signaling

Signaling protocol and handlers WebRTC signaling protocol

transport

ant-quic transport integration Transport layer implementations

types

Core WebRTC types and data structures WebRTC types and data structures