diff --git a/README.md b/README.md index eec4e20..70eb666 100644 --- a/README.md +++ b/README.md @@ -1,55 +1,20 @@ # 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 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)