lws-webrtc

This plugin implements a shared WebRTC signaling and media transport layer within libwebsockets. It handles the WebSocket signaling, DTLS (Datagram Transport Layer Security) negotiation, SRTP (Secure Real-time Transport Protocol) keying, and RTP packetization for both audio and video streams.

Features

• Signaling: Parses and generates SDP (Session Description Protocol) offer/answer exchanges over a secure WebSocket connection.
• ICE Handling: Processes STUN (Session Traversal Utilities for NAT) binding requests to establish connectivity without full ICE-agent complexity (acts as an ICE-lite server).
• Security: Sets up DTLS handshakes and extracts SRTP keys to encrypt outgoing RTP media flows.
• Media Delivery: Provides a C API (lws_webrtc_send_video, lws_webrtc_send_audio) to inject raw H.264/AV1 NAL units and Opus frames into the WebRTC session.
• Packetization: Automatically fragments H.264/AV1 bitstreams and wraps them in RTP packets suitable for browsers.
• Feedback: Receives and processes RTCP (RTP Control Protocol) feedback, particularly NACKs and PLIs (Picture Loss Indications), to maintain stream health.

Usage

This protocol is usually utilized as an underlying infrastructure plugin for higher-level media handlers (like protocol_lws_rtc_camera or protocol_lws_webrtc_mixer), which feed the raw media frames into lws_webrtc.

It provides operations via lws_vhost_name_to_protocol(vh, "lws-webrtc")->user, exposing functions like:

• lws_webrtc_send_video()
• lws_webrtc_send_audio()
• lws_webrtc_send_text()
• Session iteration and telemetry functions.

When a client connects to the lws-webrtc WebSocket endpoint, the server responds with an SDP offer containing the supported codecs (H.264, AV1, Opus) and negotiates the WebRTC UDP data path transparently.