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webrtc: refactor fixes

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This commit is contained in:
Koushik Dutta
2022-09-14 20:10:34 -07:00
parent 0ad0e6765b
commit 148da12eb5
7 changed files with 25 additions and 16 deletions
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2
plugins/webrtc/.vscode/settings.json vendored
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@@ -1,4 +1,4 @@
{
"scrypted.debugHost": "127.0.0.1",
"scrypted.debugHost": "koushik-mac",
}

4
plugins/webrtc/package-lock.json generated
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@@ -1,12 +1,12 @@
{
"name": "@scrypted/webrtc",
"version": "0.0.68",
"version": "0.0.69",
"lockfileVersion": 2,
"requires": true,
"packages": {
"": {
"name": "@scrypted/webrtc",
"version": "0.0.68",
"version": "0.0.69",
"dependencies": {
"@koush/werift": "file:../../external/werift/packages/webrtc",
"@scrypted/common": "file:../../common",

2
plugins/webrtc/package.json
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@@ -1,6 +1,6 @@
{
"name": "@scrypted/webrtc",
"version": "0.0.68",
"version": "0.0.69",
"scripts": {
"scrypted-setup-project": "scrypted-setup-project",
"prescrypted-setup-project": "scrypted-package-json",

4
plugins/webrtc/src/ffmpeg-to-wrtc.ts
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@@ -400,9 +400,11 @@ export class WebRTCConnectionManagement implements RTCConnectionManagement {
const ret = new WebRTCTrack(this, videoTransceiver, audioTransceiver, this.options?.disableIntercom ? undefined : intercom);
this.negotiation.then(async () => {
this.console.log('waiting ice connected');
await waitIceConnected(this.pc);
if (ret.removed.finished)
return;
await waitIceConnected(this.pc);
this.console.log('done waiting ice connected');
const f = await createTrackForwarder(videoTransceiver, audioTransceiver);
waitClosed(this.pc).finally(() => f.kill());
ret.removed.promise.finally(() => f.kill());

24
plugins/webrtc/src/main.ts
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@@ -6,12 +6,13 @@ import { createBrowserSignalingSession } from "@scrypted/common/src/rtc-connect"
import { connectRTCSignalingClients } from '@scrypted/common/src/rtc-signaling';
import { StorageSettings } from '@scrypted/common/src/settings';
import { SettingsMixinDeviceBase, SettingsMixinDeviceOptions } from '@scrypted/common/src/settings-mixin';
import { sleep } from '@scrypted/common/src/sleep';
import sdk, { BufferConverter, BufferConvertorOptions, DeviceCreator, DeviceCreatorSettings, DeviceProvider, FFmpegInput, HttpRequest, Intercom, MediaObject, MixinProvider, RequestMediaStream, RequestMediaStreamOptions, ResponseMediaStreamOptions, RTCAVSignalingSetup, RTCSessionControl, RTCSignalingChannel, RTCSignalingSession, ScryptedDeviceType, ScryptedInterface, ScryptedMimeTypes, Setting, Settings, SettingValue, VideoCamera } from '@scrypted/sdk';
import crypto from 'crypto';
import net from 'net';
import { DataChannelDebouncer } from './datachannel-debouncer';
import { createRTCPeerConnectionSink, parseOptions, RTC_BRIDGE_NATIVE_ID, WebRTCBridge, WebRTCConnectionManagement } from "./ffmpeg-to-wrtc";
import { stunIceServers } from './ice-servers';
import { stunIceServers, stunServer } from './ice-servers';
import { waitClosed, waitConnected, waitIceConnected } from './peerconnection-util';
import { WebRTCCamera } from "./webrtc-camera";
import { WeriftSignalingSession } from './werift-signaling-session';
@@ -301,7 +302,7 @@ export class WebRTCPlugin extends AutoenableMixinProvider implements DeviceCreat
}
async onConnection(request: HttpRequest, webSocketUrl: string) {
const cleanup = new Deferred<void>();
const cleanup = new Deferred<string>();
cleanup.promise.catch(e => this.console.log('cleaning up rtc connection:', e.message));
const ws = new WebSocket(webSocketUrl);
@@ -325,7 +326,7 @@ export class WebRTCPlugin extends AutoenableMixinProvider implements DeviceCreat
const close = () => {
const str = 'Connection closed while waiting for message';
reject(new Error(str));
cleanup.reject(new Error(str));
cleanup.resolve(str);
};
ws.addEventListener('close', close);
@@ -343,7 +344,11 @@ export class WebRTCPlugin extends AutoenableMixinProvider implements DeviceCreat
this.storageSettings.values.maximumCompatibilityMode, transcodeWidth, sessionSupportsH264High, {
setup: {
configuration: {
iceServers: stunIceServers,
iceServers: [
// seemingly this is faster than google which may have throttling on requests?
// unsure, but leaving this as is.
stunServer,
],
},
}
});
@@ -358,17 +363,18 @@ export class WebRTCPlugin extends AutoenableMixinProvider implements DeviceCreat
const { pc } = connection;
const dc = pc.createDataChannel('rpc');
waitClosed(pc).then(() => cleanup.reject(new Error('peer connection closed')));
waitClosed(pc).then(() => cleanup.resolve('peer connection closed'));
const start = Date.now();
await connection.negotiateRTCSignalingSession();
await waitIceConnected(pc);
// await waitIceConnected(pc);
// await sleep(5000);
// const [dc] = await dcPromise;
dc.message.subscribe(message => socket.write(message));
const cp = await client.clientPromise;
cp.on('close', () => cleanup.reject(new Error('socket client closed')));
cp.on('close', () => cleanup.resolve('socket client closed'));
process.send(message, cp);
const debouncer = new DataChannelDebouncer({
@@ -378,11 +384,11 @@ export class WebRTCPlugin extends AutoenableMixinProvider implements DeviceCreat
socket.destroy();
});
socket.on('data', data => debouncer.send(data));
socket.on('close', () => cleanup.reject(new Error('socket closed')));
socket.on('close', () => cleanup.resolve('socket closed'));
}
catch (e) {
console.error("error negotiating browser RTCC signaling", e);
cleanup.reject(e);
cleanup.resolve('error');
throw e;
}
}

4
plugins/webrtc/src/peerconnection-util.ts
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@@ -31,7 +31,7 @@ export function waitClosed(pc: RTCPeerConnection) {
console.log('iceGatheringStateChange', pc.iceGatheringState);
});
pc.iceConnectionStateChange.subscribe(() => {
console.log('iceConnectionStateChange', pc.connectionState, pc.iceConnectionState);
console.log('iceConnectionStateChange', pc.iceConnectionState);
if (pc.iceConnectionState === 'disconnected'
|| pc.iceConnectionState === 'failed'
|| pc.iceConnectionState === 'closed') {
@@ -39,7 +39,7 @@ export function waitClosed(pc: RTCPeerConnection) {
}
});
pc.connectionStateChange.subscribe(() => {
console.log('connectionStateChange', pc.connectionState, pc.iceConnectionState);
console.log('connectionStateChange', pc.iceConnectionState);
if (pc.connectionState === 'closed'
|| pc.connectionState === 'disconnected'
|| pc.connectionState === 'failed') {

1
plugins/webrtc/src/werift-util.ts
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@@ -43,6 +43,7 @@ export function getWeriftIceServers(configuration: RTCConfiguration): RTCIceServ
}
export function logIsPrivateIceTransport(console: Console, pc: RTCPeerConnection) {
return true;
let isPrivate = true;
for (const ice of pc.iceTransports) {
const [address, port] = ice.connection.remoteAddr;