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# 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. It aims at being both straightforward and lightweight with a minimum of external dependencies, to enable direct connectivity between native applications and web browsers without the pain of importing Google's bloated [reference library](https://webrtc.googlesource.com/src/). 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.
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 a minimum of external dependencies, to enable direct connectivity between native applications and web browsers without the pain of importing Google's bloated [reference library](https://webrtc.googlesource.com/src/). 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](https://www.openssl.org/) or [GnuTLS](https://www.gnutls.org/).
- The connectivity for WebRTC can be provided through my ad-hoc ICE library [libjuice](https://github.com/paullouisageneau/libjuice) as submodule or through [libnice](https://github.com/libnice/libnice).
The WebRTC stack is fully compatible with Firefox and Chromium, see [Compatibility](#Compatibility) below.
Licensed under LGPLv2, see [LICENSE](https://github.com/paullouisageneau/libdatachannel/blob/master/LICENSE).
## Compatibility
The library implements the following communication protocols:
### WebRTC Data Channels and Media Transport
The WebRTC stack has been tested to be compatible with Firefox and Chromium.
Protocol stack:
- SCTP-based Data Channels ([RFC8831](https://tools.ietf.org/html/rfc8831))
- SRTP-based Media Transport ([RFC8834](https://tools.ietf.org/html/rfc8834))
- DTLS/UDP ([RFC7350](https://tools.ietf.org/html/rfc7350) and [RFC8261](https://tools.ietf.org/html/rfc8261))
- ICE ([RFC8445](https://tools.ietf.org/html/rfc8445)) with STUN ([RFC8489](https://tools.ietf.org/html/rfc8489)) and its extension TURN ([RFC8656](https://tools.ietf.org/html/rfc8656))
Features:
- Full IPv6 support (as mandated by [RFC8835](https://tools.ietf.org/html/rfc8835))
- Trickle ICE ([RFC8838](https://tools.ietf.org/html/rfc8838))
- JSEP-compatible session establishment with SDP ([RFC8829](https://tools.ietf.org/html/rfc8829))
- SCTP over DTLS with SDP offer/answer ([RFC8841](https://tools.ietf.org/html/rfc8841))
- DTLS with ECDSA or RSA keys ([RFC8824](https://tools.ietf.org/html/rfc8827))
- SRTP and SRTCP key derivation from DTLS ([RFC5764](https://tools.ietf.org/html/rfc5764))
- Multicast DNS candidates ([draft-ietf-rtcweb-mdns-ice-candidates-04](https://tools.ietf.org/html/draft-ietf-rtcweb-mdns-ice-candidates-04))
- Differentiated Services QoS ([draft-ietf-tsvwg-rtcweb-qos-18](https://tools.ietf.org/html/draft-ietf-tsvwg-rtcweb-qos-18))
Note only SDP BUNDLE mode is supported for media multiplexing ([RFC8843](https://tools.ietf.org/html/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](https://tools.ietf.org/html/rfc5761)) and SCTP/DTLS data traffic ([RFC8261](https://tools.ietf.org/html/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](https://tools.ietf.org/html/rfc6455)), client-side only
- HTTP over TLS ([RFC2818](https://tools.ietf.org/html/rfc2818))
Features:
- IPv6 and IPv4/IPv6 dual-stack support
- Keepalive with ping/pong
## Dependencies
Only [GnuTLS](https://www.gnutls.org/) or [OpenSSL](https://www.openssl.org/) are necessary.
Optionally, [libnice](https://nice.freedesktop.org/) can be selected as an alternative ICE backend instead of libjuice.
Only [GnuTLS](https://www.gnutls.org/) or [OpenSSL](https://www.openssl.org/) are necessary. Optionally, [libnice](https://nice.freedesktop.org/) can be selected as an alternative ICE backend instead of libjuice.
Submodules:
- libjuice: https://github.com/paullouisageneau/libjuice
@ -153,6 +118,42 @@ ws.onMessage([](std::variant<binary, string> message) {
ws.open("wss://my.websocket/service");
```
## Compatibility
The library implements the following communication protocols:
### WebRTC Data Channels and Media Transport
Protocol stack:
- SCTP-based Data Channels ([RFC8831](https://tools.ietf.org/html/rfc8831))
- SRTP-based Media Transport ([RFC8834](https://tools.ietf.org/html/rfc8834))
- DTLS/UDP ([RFC7350](https://tools.ietf.org/html/rfc7350) and [RFC8261](https://tools.ietf.org/html/rfc8261))
- ICE ([RFC8445](https://tools.ietf.org/html/rfc8445)) with STUN ([RFC8489](https://tools.ietf.org/html/rfc8489)) and its extension TURN ([RFC8656](https://tools.ietf.org/html/rfc8656))
Features:
- Full IPv6 support (as mandated by [RFC8835](https://tools.ietf.org/html/rfc8835))
- Trickle ICE ([RFC8838](https://tools.ietf.org/html/rfc8838))
- JSEP-compatible session establishment with SDP ([RFC8829](https://tools.ietf.org/html/rfc8829))
- SCTP over DTLS with SDP offer/answer ([RFC8841](https://tools.ietf.org/html/rfc8841))
- DTLS with ECDSA or RSA keys ([RFC8824](https://tools.ietf.org/html/rfc8827))
- SRTP and SRTCP key derivation from DTLS ([RFC5764](https://tools.ietf.org/html/rfc5764))
- Multicast DNS candidates ([draft-ietf-rtcweb-mdns-ice-candidates-04](https://tools.ietf.org/html/draft-ietf-rtcweb-mdns-ice-candidates-04))
- Differentiated Services QoS ([draft-ietf-tsvwg-rtcweb-qos-18](https://tools.ietf.org/html/draft-ietf-tsvwg-rtcweb-qos-18))
Note only SDP BUNDLE mode is supported for media multiplexing ([RFC8843](https://tools.ietf.org/html/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](https://tools.ietf.org/html/rfc5761)) and SCTP/DTLS data traffic ([RFC8261](https://tools.ietf.org/html/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](https://tools.ietf.org/html/rfc6455)), client-side only
- HTTP over TLS ([RFC2818](https://tools.ietf.org/html/rfc2818))
Features:
- IPv6 and IPv4/IPv6 dual-stack support
- Keepalive with ping/pong
## External resources
- Rust wrapper for libdatachannel: [datachannel-rs](https://github.com/lerouxrgd/datachannel-rs)
- Node.js wrapper for libdatachannel: [node-datachannel](https://github.com/murat-dogan/node-datachannel)