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Add more debug methods to WebRTC module

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sergystepanov 2026-06-01 12:12:26 +03:00
parent ee56aba302
commit 440102fc1b
1 changed files with 41 additions and 21 deletions
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62
web/js/network/webrtc.js
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@ -23,7 +23,7 @@ let inputReady = false;
let onData;
const start = (iceservers) => {
log.info('[rtc] <- ICE servers', iceservers);
log.debug('[rtc] <- ICE servers', iceservers);
const servers = iceservers || [];
connection = new RTCPeerConnection({iceServers: servers});
mediaStream = new MediaStream();
@ -44,7 +44,7 @@ const start = (iceservers) => {
dataChannel = e.channel;
dataChannel.onopen = () => {
log.info('[rtc] the input channel has been opened');
log.debug('[rtc] the input channel has been opened');
inputReady = true;
pub(WEBRTC_CONNECTION_READY)
};
@ -53,12 +53,16 @@ const start = (iceservers) => {
}
dataChannel.onclose = () => {
inputReady = false
log.info('[rtc] the input channel has been closed')
log.debug('[rtc] the input channel has been closed')
}
}
connection.oniceconnectionstatechange = ice.onIceConnectionStateChange;
connection.onicegatheringstatechange = ice.onIceStateChange;
connection.onicecandidate = ice.onIcecandidate;
connection.onicecandidate = ice.onIceCandidate;
connection.onicecandidateerror = ice.onIceCandidateError;
connection.onconnectionstatechange = _ => {
console.debug(`[rtc] connection state -> ${connection.connectionState}`)
}
connection.ontrack = event => {
mediaStream.addTrack(event.track);
}
@ -93,33 +97,36 @@ const stop = () => {
}
const ice = (() => {
const ICE_TIMEOUT = 2000;
const ICE_TIMEOUT = 3000;
let timeForIceGathering;
return {
onIcecandidate: data => {
onIceCandidate: data => {
if (!data.candidate) return;
log.info('[rtc] user candidate', data.candidate);
log.debug('[rtc] user candidate', data.candidate);
pub(WEBRTC_ICE_CANDIDATE_FOUND, {candidate: data.candidate})
},
onIceCandidateError: event => {
log.debug('[rtc] ice candidate error', event)
},
onIceStateChange: event => {
switch (event.target.iceGatheringState) {
case 'gathering':
log.info('[rtc] ice gathering');
log.debug('[rtc] ice gathering');
timeForIceGathering = setTimeout(() => {
log.warn(`[rtc] ice gathering was aborted due to timeout ${ICE_TIMEOUT}ms`);
// sendCandidates();
}, ICE_TIMEOUT);
break;
case 'complete':
log.info('[rtc] ice gathering has been completed');
log.debug('[rtc] ice gathering has been completed');
if (timeForIceGathering) {
clearTimeout(timeForIceGathering);
}
}
},
onIceConnectionStateChange: () => {
log.info('[rtc] <- iceConnectionState', connection.iceConnectionState);
log.debug('[rtc] <- iceConnectionState', connection.iceConnectionState);
switch (connection.iceConnectionState) {
case 'connected':
log.info('[rtc] connected...');
@ -151,19 +158,32 @@ export const webrtc = {
start,
setRemoteDescription: async (data, media) => {
log.debug('[rtc] remote SDP', data)
const offer = new RTCSessionDescription(JSON.parse(atob(data)));
await connection.setRemoteDescription(offer);
const decodedSDP = JSON.parse(atob(data))
const offer = new RTCSessionDescription(decodedSDP);
const answer = await connection.createAnswer();
// Chrome bug https://bugs.chromium.org/p/chromium/issues/detail?id=818180 workaround
// force stereo params for Opus tracks (a=fmtp:111 ...)
answer.sdp = answer.sdp.replace(/(a=fmtp:111 .*)/g, '$1;stereo=1');
await connection.setLocalDescription(answer);
log.debug("[rtc] local SDP", answer)
try {
await connection.setRemoteDescription(offer);
} catch (e) {
log.error('[rtc] remote SDP error', e)
}
log.debug(`[rtc] remote Trickle ICE support: ${connection.canTrickleIceCandidates}`)
try {
const answer = await connection.createAnswer();
// Chrome bug https://bugs.chromium.org/p/chromium/issues/detail?id=818180 workaround
// force stereo params for Opus tracks (a=fmtp:111 ...)
answer.sdp = answer.sdp.replace(/(a=fmtp:111 .*)/g, '$1;stereo=1');
await connection.setLocalDescription(answer);
log.debug("[rtc] local SDP", answer)
isAnswered = true;
pub(WEBRTC_ICE_CANDIDATES_FLUSH);
pub(WEBRTC_SDP_ANSWER, {sdp: answer});
} catch (e) {
log.error('[rtc] answer/local SDP error', e)
}
isAnswered = true;
pub(WEBRTC_ICE_CANDIDATES_FLUSH);
pub(WEBRTC_SDP_ANSWER, {sdp: answer});
media.srcObject = mediaStream;
},
addCandidate: (data) => {