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webrtc: fix race condition in address gathering

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This commit is contained in:
Koushik Dutta
2023-01-09 21:43:33 -08:00
parent 214a13183e
commit a85ac600eb
3 changed files with 12 additions and 9 deletions
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4
plugins/webrtc/package-lock.json generated
View File

@@ -1,12 +1,12 @@
{
"name": "@scrypted/webrtc",
"version": "0.1.3",
"version": "0.1.18",
"lockfileVersion": 2,
"requires": true,
"packages": {
"": {
"name": "@scrypted/webrtc",
"version": "0.1.3",
"version": "0.1.18",
"dependencies": {
"@scrypted/common": "file:../../common",
"@scrypted/sdk": "file:../../sdk",

2
plugins/webrtc/package.json
View File

@@ -1,6 +1,6 @@
{
"name": "@scrypted/webrtc",
"version": "0.1.3",
"version": "0.1.18",
"scripts": {
"scrypted-setup-project": "scrypted-setup-project",
"prescrypted-setup-project": "scrypted-package-json",

15
plugins/webrtc/src/main.ts
View File

@@ -72,14 +72,14 @@ class WebRTCMixin extends SettingsMixinDeviceBase<RTCSignalingClient & VideoCame
if ((this.type === ScryptedDeviceType.Speaker || this.type === ScryptedDeviceType.SmartSpeaker)
&& this.mixinDeviceInterfaces.includes(ScryptedInterface.RTCSignalingChannel)) {
this.console.log('starting webrtc speaker intercom');
this.console.log('starting webrtc speaker intercom');
const pc = new RTCPeerConnection();
const atrack = new MediaStreamTrack({ kind: 'audio' });
const audioTransceiver = pc.addTransceiver(atrack);
const weriftSignalingSession = new WeriftSignalingSession(this.console, pc);
const control = await this.mixinDevice.startRTCSignalingSession(weriftSignalingSession);
const forwarder = await createTrackForwarder({
timeStart: Date.now(),
videoTransceiver: undefined,
@@ -509,7 +509,7 @@ export class WebRTCPlugin extends AutoenableMixinProvider implements DeviceCreat
export async function fork() {
return {
async createConnection(message: any, port: number, clientSession: RTCSignalingSession, maximumCompatibilityMode: boolean, transcodeWidth: number, sessionSupportsH264High: boolean, options: { disableIntercom?: boolean; configuration: RTCConfiguration, weriftConfiguration: PeerConfig; }) {
async createConnection(message: any, portOrDummy: number | boolean, clientSession: RTCSignalingSession, maximumCompatibilityMode: boolean, transcodeWidth: number, sessionSupportsH264High: boolean, options: { disableIntercom?: boolean; configuration: RTCConfiguration, weriftConfiguration: PeerConfig; }) {
const cleanup = new Deferred<string>();
cleanup.promise.catch(e => this.console.log('cleaning up rtc connection:', e.message));
cleanup.promise.finally(() => setTimeout(() => process.exit(), 10000));
@@ -529,7 +529,8 @@ export async function fork() {
}
}
if (port) {
if (typeof portOrDummy === 'number') {
const port = portOrDummy;
const socket = net.connect(port, '127.0.0.1');
cleanup.promise.finally(() => socket.destroy());
@@ -548,8 +549,10 @@ export async function fork() {
}
else {
pc.createDataChannel('dummy');
const offer = await pc.createOffer();
pc.setLocalDescription(offer);
if (portOrDummy) {
const offer = await pc.createOffer();
pc.setLocalDescription(offer);
}
}
return connection;