werift-webrtc

To improve compatibility with browsers, undefined candidates are now included in the payload of events such as peerConnection.onIceCandidate when ice gathering is completed.

Move nonstandard module to subpath.
When using subpath, set moduleResolution in tsconfig to node16 or nodenext

• exclude all inactive descriptions from bundle answer #339

• improve recording implementation #259
• Fix GenericNack serialization #261
• bundled media should use any available iceParams from any other bundled media sections #264
• add support for stop/remove tracks. #250
• replace rtp option to signal the stream contains a discontunity #267

• renegotiation should reset the sequence numbers to 0 #255
• Add function to filter STUN responses #257
• Add iceUseIpv4/6 config to RTCPeerConnection #252
• Allow use of specific network interface #249
• Feature/improve reco #259
  • rewrite RTP processor
  • Added MediaRecorder option

#230

          const pc = new RTCPeerConnection({
            codecs: {
              audio: [
                new RTCRtpCodecParameters({
                  mimeType: "PCMU",
                  clockRate: 8000,
                  channels: 1,
                  payloadType: 0
                }),
              ],
            },
          });

How to specify BundlePolicy

const pc = new RTCPeerConnection({
            bundlePolicy: "disable" or "max-compat" or "max-bundle",
          });

Default value is max-compat

Example code is here
https://github.com/shinyoshiaki/werift-webrtc/blob/develop/examples/save_to_disk/av1x.ts

Video and Audio can be saved to a webm file using Node.js alone, without the need for gStreamer.

example

import { MediaRecorder, RTCPeerConnection } from "werift";
import { Server } from "ws";

// open ./answer.html

const server = new Server({ port: 8888 });
console.log("start");

server.on("connection", async (socket) => {
  const recorder = new MediaRecorder([], "./test.webm", {
    width: 640,
    height: 360,
  });

  const pc = new RTCPeerConnection();

  {
    const transceiver = pc.addTransceiver("video");

    transceiver.onTrack.subscribe((track) => {
      recorder.addTrack(track);
      if (recorder.tracks.length === 2) {
        recorder.start();
      }
      setInterval(() => {
        transceiver.receiver.sendRtcpPLI(track.ssrc);
      }, 15_000);
    });
  }
  {
    const transceiver = pc.addTransceiver("audio");
    transceiver.onTrack.subscribe((track) => {
      recorder.addTrack(track);
      if (recorder.tracks.length === 2) {
        recorder.start();
      }
    });
  }

  await pc.setLocalDescription(await pc.createOffer());
  const sdp = JSON.stringify(pc.localDescription);
  socket.send(sdp);

  socket.on("message", (data: any) => {
    pc.setRemoteDescription(JSON.parse(data));
  });
});