#WebRTC #networking #p2p #tcp #udp #rtcpeerconnection

tytan652 2dc3d77e07 Added USE_SYSTEM_JSON cmake option 2 yıl önce
.github fc0eb3bf4d Added build-nomedia workflow 3 yıl önce
cmake 7aced2860c Match Upstream LibSRTP CMake Targets 3 yıl önce
deps 791d5204b1 Updated libsrtp to v2.5.0 2 yıl önce
examples 2dc3d77e07 Added USE_SYSTEM_JSON cmake option 2 yıl önce
include 43edfa669b Provide ability to bind WebSocketServer to specifc address 2 yıl önce
pages f31b861c63 Typo 2 yıl önce
src d16c25058f Merge pull request #812 from web2098/AllowEmptyValueHeaders 2 yıl önce
test 93a732561d Added test for rtcReceiveMessage() 2 yıl önce
.clang-format 17f99252cd clang-format does not support python 5 yıl önce
.editorconfig ea8cd06964 Converted web example to WebSocket signaling 5 yıl önce
.gitignore b592e5fc09 Updated .gitignore for Python 4 yıl önce
.gitmodules 4c3b0b2f11 Mark external dependency submodules as shallow to speed up the cloning process 3 yıl önce
.travis.yml 5c12b25717 Added .travis.yml for FreeBSD build 3 yıl önce
BUILDING.md f2bd0897a0 Building: Enhanced info and examples 3 yıl önce
CMakeLists.txt 2dc3d77e07 Added USE_SYSTEM_JSON cmake option 2 yıl önce
DOC.md 7c75864ea7 Updated doc 2 yıl önce
Jamfile 9785b2724e Jamfile: Fixed static linking flags for libjuice with MSVC 2 yıl önce
LICENSE a0f3f01bf5 Re-licensed under MPL 2.0 2 yıl önce
Makefile 62f1def192 Added RTC_EXPORTS define to Makefile for consistency 2 yıl önce
README.md 7b3e404d0c Readme: Fixed license badge link 2 yıl önce

README.md

libdatachannel - C/C++ WebRTC network library

Build with OpenSSL Build with GnuTLS Gitter Discord

AUR package FreeBSD port Vcpkg package

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, FreeBSD, Apple macOS and iOS) and Microsoft Windows. WebRTC is a W3C and IETF standard enabling real-time peer-to-peer data and media exchange between two devices.

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 browsers like Firefox and Chromium, see Compatibility below. Additionally, code using Data Channels and WebSockets from the library may be compiled as is to WebAssembly for browsers with datachannel-wasm.

libdatachannel is licensed under MPL 2.0 since version 0.18, see LICENSE (previous versions were licensed under LGPLv2.1 or later).

libdatachannel is available on AUR, vcpkg, and FreeBSD ports. Bindings are available for Rust and Node.js.

Dependencies

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

Submodules:

Building

See BUILDING.md for building instructions.

Examples

See examples for complete usage examples with signaling server (under MPL 2.0).

Additionally, 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::PeerConnection pc(config);

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

MY_ON_RECV_CANDIDATE_FROM_REMOTE([&pc](std::string candidate, std::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<rtc::binary, rtc::string> message) {
    if (std::holds_alternative<rtc::string>(message)) {
        std::cout << "Received: " << get<rtc::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<rtc::binary, rtc::string> message) {
    if (std::holds_alternative<rtc::string>(message)) {
        std::cout << "WebSocket received: " << std::get<rtc::string>(message) << endl;
    }
});

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

Compatibility

The library implements the following communication protocols:

WebRTC Data Channels and Media Transport

WebRTC allows real-time data and media exchange between two devices through a Peer Connection (or RTCPeerConnection), a signaled peer-to-peer connection which can carry both Data Channels and media tracks. It is compatible with browsers Firefox, Chromium, and Safari, and other WebRTC libraries (see webrtc-echoes). Media transport is optional and can be disabled at compile time.

Protocol stack:

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 (RFC8827)
  • SRTP and SRTCP key derivation from DTLS (RFC5764)
  • Differentiated Services QoS (RFC8837) where possible
  • Multicast DNS candidates (draft-ietf-rtcweb-mdns-ice-candidates-04)
  • Multiplexing connections on a single UDP port with libjuice as ICE backend

Note only SDP BUNDLE mode is supported for media multiplexing (RFC8843). The behavior is equivalent to the JSEP bundle-only policy: the library always negotiates 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.

Protocol stack:

  • WebSocket protocol (RFC6455), client and server side
  • HTTP over TLS (RFC2818)

Features:

  • IPv6 and IPv4/IPv6 dual-stack support
  • Keepalive with ping/pong

External resources

Thanks

Thanks to Streamr, Vagon, Deon Botha, and Michael Cho for sponsoring this work!