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webrtc: fix unhandled rejection

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This commit is contained in:
Koushik Dutta
2024-03-16 11:51:17 -07:00
parent 374c5364f4
commit 1fa2cae936
1 changed files with 26 additions and 11 deletions
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37
plugins/webrtc/src/wrtc-to-rtsp.ts
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@@ -9,7 +9,7 @@ import sdk, { FFmpegInput, Intercom, MediaObject, MediaStreamUrl, ResponseMediaS
import { logConnectionState, waitClosed, waitConnected, waitIceConnected } from "./peerconnection-util";
import { startRtpForwarderProcess } from "./rtp-forwarders";
import { getFFmpegRtpAudioOutputArguments, requiredAudioCodecs, requiredVideoCodec } from "./webrtc-required-codecs";
import { createRawResponse, getWeriftIceServers, isPeerConnectionAlive } from "./werift-util";
import { createRawResponse, getWeriftIceServers, isPeerConnectionAlive, logIsLocalIceTransport } from "./werift-util";
import { H264Repacketizer } from "../../homekit/src/types/camera/h264-packetizer";
const { mediaManager } = sdk;
@@ -20,6 +20,14 @@ export interface RTCPeerConnectionPipe {
pcClose: Promise<unknown>;
}
function ignoreDeferred(...d: Deferred<any>[]) {
d.forEach(d => d.promise.catch(() => { }));
}
function ignorePromise(...p: Promise<any>[]) {
p.forEach(p => p.catch(() => { }));
}
export async function createRTCPeerConnectionSource(options: {
console: Console,
mediaStreamOptions: ResponseMediaStreamOptions,
@@ -35,13 +43,14 @@ export async function createRTCPeerConnectionSource(options: {
const sessionControl = new Deferred<RTCSessionControl>();
const peerConnection = new Deferred<RTCPeerConnection>();
const intercom = new Deferred<Intercom>();
ignoreDeferred(sessionControl, intercom);
const cleanup = () => {
console.log('webrtc/rtsp cleaning up');
clientPromise.then(client => client.destroy());
sessionControl.promise.then(sc => sc.endSession());
peerConnection.promise.then(pc => pc.close());
intercom.promise.then(intercom => intercom.stopIntercom());
clientPromise.then(client => client.destroy()).catch(() => {});
sessionControl.promise.then(sc => sc.endSession()).catch(() => {});
peerConnection.promise.then(pc => pc.close()).catch(() => {});
ignorePromise(intercom.promise.then(intercom => intercom.stopIntercom()));
};
clientPromise.then(socket => socket.on('close', cleanup));
@@ -69,6 +78,18 @@ export async function createRTCPeerConnectionSource(options: {
logConnectionState(console, ret);
peerConnection.resolve(ret);
(async () => {
try {
if (ret.remoteIsBundled)
await waitConnected(ret);
else
await waitIceConnected(ret);
logIsLocalIceTransport(console, ret);
}
catch (e) {
}
})();
}
let audioTrack: string;
@@ -82,12 +103,6 @@ export async function createRTCPeerConnectionSource(options: {
let gotVideo = false;
const pc = await peerConnection.promise;
pc.iceConnectionStateChange.subscribe(() => {
console.log('iceConnectionState', pc.iceConnectionState);
});
pc.connectionStateChange.subscribe(() => {
console.log('connectionState', pc.connectionState);
});
const setupAudioTranscevier = (transciever: RTCRtpTransceiver) => {
audioTransceiver = transciever;