webrtc: reduce debug logging

common/src/rtc-signaling.ts

@@ -250,7 +250,8 @@ export class BrowserSignalingSession implements RTCSignalingSession {
 function logSendCandidate(console: Console, type: string, session: RTCSignalingSession): RTCSignalingSendIceCandidate {
     return async (candidate) => {
         try {
-            console.log(`${type} trickled candidate:`, candidate.sdpMLineIndex, candidate.candidate);
+            if (localStorage.getItem('debugLog') === 'true')
+                console.log(`${type} trickled candidate:`, candidate.sdpMLineIndex, candidate.candidate);
             await session.addIceCandidate(candidate);
         }
         catch (e) {
@@ -308,11 +309,13 @@ export async function connectRTCSignalingClients(
 
     const offer = await offerClient.createLocalDescription('offer', offerSetup as RTCAVSignalingSetup,
         disableTrickle ? undefined : answerQueue.queueSendCandidate);
-    console.log('offer sdp', offer.sdp);
+    if (localStorage.getItem('debugLog') === 'true')
+        console.log('offer sdp', offer.sdp);
     await answerClient.setRemoteDescription(offer, answerSetup as RTCAVSignalingSetup);
     const answer = await answerClient.createLocalDescription('answer', answerSetup as RTCAVSignalingSetup,
         disableTrickle ? undefined : offerQueue.queueSendCandidate);
-    console.log('answer sdp', answer.sdp);
+    if (localStorage.getItem('debugLog') === 'true')
+        console.log('answer sdp', answer.sdp);
     await offerClient.setRemoteDescription(answer, offerSetup as RTCAVSignalingSetup);
     offerQueue.flush();
     answerQueue.flush();

plugins/webrtc/package.json

@@ -1,6 +1,6 @@
 {
    "name": "@scrypted/webrtc",
-   "version": "0.1.38",
+   "version": "0.1.39",
    "scripts": {
       "scrypted-setup-project": "scrypted-setup-project",
       "prescrypted-setup-project": "scrypted-package-json",

plugins/webrtc/src/ffmpeg-to-wrtc.ts

@@ -4,7 +4,7 @@ import { Deferred } from "@scrypted/common/src/deferred";
 import sdk, { FFmpegInput, FFmpegTranscodeStream, Intercom, MediaObject, MediaStreamDestination, MediaStreamFeedback, RequestMediaStream, RTCAVSignalingSetup, RTCConnectionManagement, RTCMediaObjectTrack, RTCSignalingOptions, RTCSignalingSession, ScryptedDevice, ScryptedMimeTypes } from "@scrypted/sdk";
 import { ScryptedSessionControl } from "./session-control";
 import { requiredAudioCodecs, requiredVideoCodec } from "./webrtc-required-codecs";
-import { logIsPrivateIceTransport } from "./werift-util";
+import { isPrivateIceTransport, logIsPrivateIceTransport } from "./werift-util";
 
 import { addVideoFilterArguments } from "@scrypted/common/src/ffmpeg-helpers";
 import { connectRTCSignalingClients } from "@scrypted/common/src/rtc-signaling";
@@ -431,10 +431,6 @@ export class WebRTCConnectionManagement implements RTCConnectionManagement {
         waitConnected(this.pc)
             .then(() => logIsPrivateIceTransport(this.console, this.pc)).catch(() => {});
 
-        this.pc.signalingStateChange.subscribe(() => {
-            this.console.log('sig change', this.pc.signalingState);
-        })
-
         this.weriftSignalingSession = new WeriftSignalingSession(console, this.pc);
     }
 
@@ -469,7 +465,7 @@ export class WebRTCConnectionManagement implements RTCConnectionManagement {
             createTrackForwarder: async (videoTransceiver: RTCRtpTransceiver, audioTransceiver: RTCRtpTransceiver) => {
                 const ret = await createTrackForwarder({
                     timeStart,
-                    ...logIsPrivateIceTransport(console, this.pc),
+                    ...isPrivateIceTransport(this.pc),
                     requestMediaStream,
                     videoTransceiver,
                     audioTransceiver,
@@ -582,7 +578,7 @@ export async function createRTCPeerConnectionSink(
     clientOffer = true,
 ) {
     const clientOptions = await clientSignalingSession.getOptions();
-    console.log('remote options', clientOptions);
+    // console.log('remote options', clientOptions);
 
     const connection = new WebRTCConnectionManagement(console, clientSignalingSession, maximumCompatibilityMode, clientOptions, {
         configuration,

plugins/webrtc/src/main.ts

@@ -223,6 +223,10 @@ export class WebRTCPlugin extends AutoenableMixinProvider implements DeviceCreat
             type: 'textarea',
             description: "RTCConfiguration that can be used to specify custom TURN and STUN servers. https://gist.github.com/koush/631d38ac8647a86baaac7b22d863f010",
         },
+        debugLog: {
+            title: 'Debug Log',
+            type: 'boolean',
+        }
     });
     bridge: WebRTCBridge;
     activeConnections = 0;

plugins/webrtc/src/werift-util.ts

@@ -42,20 +42,15 @@ export function getWeriftIceServers(configuration: RTCConfiguration): RTCIceServ
     return ret;
 }
 
-export function logIsPrivateIceTransport(console: Console, pc: RTCPeerConnection) {
+export function isPrivateIceTransport(pc: RTCPeerConnection) {
     let isPrivate = true;
     let destinationId: string;
     for (const ice of pc.iceTransports) {
         const [address, port] = (ice.connection as any).nominated[1].remoteAddr;
         if (!destinationId)
             destinationId = address;
-        const { turnServer } = ice.connection;
         isPrivate = isPrivate && ip.isPrivate(address);
-        console.log('ice transport ip', {
-            address,
-            port,
-            turnServer: !!turnServer,
-        });
+
     }
     console.log('Connection is local network:', isPrivate);
     const ipv4 = ip.isV4Format(destinationId);
@@ -65,3 +60,10 @@ export function logIsPrivateIceTransport(console: Console, pc: RTCPeerConnection
         destinationId,
     };
 }
+
+export function logIsPrivateIceTransport(console: Console, pc: RTCPeerConnection) {
+    const ret = isPrivateIceTransport(pc);
+    console.log('ice transport', ret);
+    console.log('Connection is local network:', ret.isPrivate);
+    return ret;
+}