libdatachannel - C/C++ WebRTC Data Channels

libdatachannel is a standalone implementation of WebRTC Data Channels, WebRTC Media Transport, and WebSockets in C++17 with C bindings for POSIX platforms (including GNU/Linux, Android, and Apple macOS) and Microsoft Windows.

The library aims at being both straightforward and lightweight with minimal external dependencies, to enable direct connectivity between native applications and web browsers without the pain of importing Google's bloated reference library. The interface consists of somewhat simplified versions of the JavaScript WebRTC and WebSocket APIs present in browsers, in order to ease the design of cross-environment applications.

It can be compiled with multiple backends:

The security layer can be provided through OpenSSL or GnuTLS.
The connectivity for WebRTC can be provided through my ad-hoc ICE library libjuice as submodule or through libnice.

The WebRTC stack is fully compatible with Firefox and Chromium, see Compatibility below.

Licensed under LGPLv2, see LICENSE.

Dependencies

Only GnuTLS or OpenSSL are necessary. Optionally, libnice can be selected as an alternative ICE backend instead of libjuice.

Submodules:

libjuice: https://github.com/paullouisageneau/libjuice
usrsctp: https://github.com/sctplab/usrsctp
libsrtp: https://github.com/cisco/libsrtp (if compiled with media support)

Building

See BUILDING.md for building instructions.

Examples

See examples for complete usage examples with signaling server (under GPLv2).

Additionnaly, you might want to have a look at the C API documentation.

Signal a PeerConnection

#include "rtc/rtc.hpp"

rtc::Configuration config;
config.iceServers.emplace_back("mystunserver.org:3478");

rtc::PeerConection pc(config);

pc.onLocalDescription([](rtc::Description sdp) {
    // Send the SDP to the remote peer
    MY_SEND_DESCRIPTION_TO_REMOTE(string(sdp));
});

pc.onLocalCandidate([](rtc::Candidate candidate) {
    // Send the candidate to the remote peer
    MY_SEND_CANDIDATE_TO_REMOTE(candidate.candidate(), candidate.mid());
});

MY_ON_RECV_DESCRIPTION_FROM_REMOTE([&pc](string sdp) {
    pc.setRemoteDescription(rtc::Description(sdp));
});

MY_ON_RECV_CANDIDATE_FROM_REMOTE([&pc](string candidate, string mid) {
    pc.addRemoteCandidate(rtc::Candidate(candidate, mid));
});

Observe the PeerConnection state

pc.onStateChange([](rtc::PeerConnection::State state) {
    std::cout << "State: " << state << std::endl;
});

pc.onGatheringStateChange([](rtc::PeerConnection::GatheringState state) {
    std::cout << "Gathering state: " << state << std::endl;
});

Create a DataChannel

auto dc = pc.createDataChannel("test");

dc->onOpen([]() {
    std::cout << "Open" << std::endl;
});

dc->onMessage([](std::variant<binary, string> message) {
    if (std::holds_alternative<string>(message)) {
        std::cout << "Received: " << get<string>(message) << std::endl;
    }
});

Receive a DataChannel

std::shared_ptr<rtc::DataChannel> dc;
pc.onDataChannel([&dc](std::shared_ptr<rtc::DataChannel> incoming) {
    dc = incoming;
    dc->send("Hello world!");
});

Open a WebSocket

rtc::WebSocket ws;

ws.onOpen([]() {
    std::cout << "WebSocket open" << std::endl;
});

ws.onMessage([](std::variant<binary, string> message) {
    if (std::holds_alternative<string>(message)) {
        std::cout << "WebSocket received: " << std::get<string>(message) << endl;
    }
});

ws.open("wss://my.websocket/service");

Compatibility

The library implements the following communication protocols:

WebRTC Data Channels and Media Transport

The library implements WebRTC Peer Connections with both Data Channels and Media Transport. Media transport is optional and can be disabled at compile time.

Protocol stack:

SCTP-based Data Channels (RFC8831)
SRTP-based Media Transport (RFC8834)
DTLS/UDP (RFC7350 and RFC8261)
ICE (RFC8445) with STUN (RFC8489) and its extension TURN (RFC8656)

Features:

Full IPv6 support (as mandated by RFC8835)
Trickle ICE (RFC8838)
JSEP-compatible session establishment with SDP (RFC8829)
SCTP over DTLS with SDP offer/answer (RFC8841)
DTLS with ECDSA or RSA keys (RFC8824)
SRTP and SRTCP key derivation from DTLS (RFC5764)
Multicast DNS candidates (draft-ietf-rtcweb-mdns-ice-candidates-04)
Differentiated Services QoS (draft-ietf-tsvwg-rtcweb-qos-18)

Note only SDP BUNDLE mode is supported for media multiplexing (RFC8843). The behavior is equivalent to the JSEP bundle-only policy: the library always negociates one unique network component, where SRTP media streams are multiplexed with SRTCP control packets (RFC5761) and SCTP/DTLS data traffic (RFC8261).

WebSocket

WebSocket is the protocol of choice for WebRTC signaling. The support is optional and can be disabled at compile time.