fix(WebRTCManager, callStore): improve signaling state handling and media track management
All checks were successful
Frontend CI / build-and-push-web (push) Successful in 2m53s
Frontend CI / ota-android (push) Successful in 11m0s
Frontend CI / build-android-apk (push) Successful in 1h27m45s

- Enhanced ontrack event handling to construct remote streams and avoid duplicate tracks during renegotiation.
- Updated negotiation logic to ensure it only occurs when the signaling state is stable and the instance is the initiator.
- Added diagnostic logging for offer and answer SDP to assist in debugging video track handling.
- Improved rollback logic in callStore to handle signaling state checks more robustly after rollback attempts.
This commit is contained in:
lafay
2026-03-28 01:41:26 +08:00
parent 7c33409624
commit fa10ef5116
2 changed files with 143 additions and 38 deletions

View File

@@ -87,11 +87,18 @@ class WebRTCManager {
// @ts-ignore
pc.ontrack = (event) => {
console.log('[WebRTC] ontrack event, kind:', event.track.kind, 'streams:', event.streams.length);
if (event.streams && event.streams[0]) {
this.remoteStream = event.streams[0];
this.emit({ type: 'remotestream', stream: event.streams[0] });
console.log('[WebRTC] ontrack event, kind:', event.track.kind, 'streams:', event.streams?.length ?? 0);
// react-native-webrtc often doesn't populate event.streams
// Manually construct the remote stream from the track
if (!this.remoteStream) {
this.remoteStream = new MediaStream();
}
// Avoid adding duplicate tracks (can happen during renegotiation)
const existingTrack = this.remoteStream.getTracks().find(t => t.id === event.track.id);
if (!existingTrack) {
this.remoteStream.addTrack(event.track);
}
this.emit({ type: 'remotestream', stream: this.remoteStream });
};
// @ts-ignore
@@ -112,7 +119,8 @@ class WebRTCManager {
// Only start negotiation if:
// 1. We're in stable state
// 2. Not already negotiating
if (pc.signalingState === 'stable' && !this.isNegotiating) {
// 3. We are the initiator (only initiator should auto-negotiate)
if (pc.signalingState === 'stable' && !this.isNegotiating && this.isInitiator) {
try {
this.isNegotiating = true;
const offer = await this.createOffer();
@@ -129,7 +137,7 @@ class WebRTCManager {
}, 500);
}
} else {
console.log('[WebRTC] Skipping negotiation: state=', pc.signalingState, 'isNegotiating=', this.isNegotiating);
console.log('[WebRTC] Skipping negotiation: state=', pc.signalingState, 'isNegotiating=', this.isNegotiating, 'isInitiator=', this.isInitiator);
}
};
@@ -238,8 +246,14 @@ class WebRTCManager {
async createOffer(): Promise<RTCSessionDescriptionInit> {
if (!this.peerConnection) throw new Error('PeerConnection not initialized');
// Check signaling state - must be stable to create offer
if (this.peerConnection.signalingState !== 'stable') {
console.warn('[WebRTC] Cannot create offer, signaling state is:', this.peerConnection.signalingState);
throw new Error(`Cannot create offer in signaling state: ${this.peerConnection.signalingState}`);
}
console.log('[WebRTC] Creating offer...');
// Check if we have video tracks
const hasLocalVideo = (this.localStream?.getVideoTracks().length ?? 0) > 0;
const offerOptions = {
@@ -250,19 +264,29 @@ class WebRTCManager {
console.log('[WebRTC] Offer options:', offerOptions, 'hasLocalVideo:', hasLocalVideo);
const offer = await this.peerConnection.createOffer(offerOptions);
// Check again after async operation
if (!this.peerConnection || this.disposed) {
throw new Error('PeerConnection was disposed during offer creation');
}
// Diagnostic: log video m-line from offer SDP
if (offer.sdp) {
const videoMLine = offer.sdp.split('\n').find((l: string) => l.startsWith('m=video'));
const videoIdx = offer.sdp.indexOf('m=video');
const videoDir = videoIdx >= 0
? offer.sdp.split('\n').find((l: string) => l.startsWith('a=') && offer.sdp.indexOf(l) > videoIdx && (l.includes('sendrecv') || l.includes('recvonly') || l.includes('sendonly') || l.includes('inactive')))
: null;
console.log('[WebRTC] Offer video m-line:', videoMLine, 'video direction:', videoDir);
}
await this.peerConnection.setLocalDescription(offer);
return offer;
}
async createAnswer(): Promise<RTCSessionDescriptionInit> {
console.log('[WebRTC] createAnswer called, peerConnection exists:', !!this.peerConnection, 'disposed:', this.disposed);
if (!this.peerConnection) {
console.error('[WebRTC] createAnswer: PeerConnection is null');
throw new Error('PeerConnection not initialized');
@@ -279,18 +303,28 @@ class WebRTCManager {
// @ts-ignore - react-native-webrtc types
const answer = await this.peerConnection.createAnswer(answerOptions);
console.log('[WebRTC] Answer created, setting local description...');
// Check again after async operation
if (!this.peerConnection || this.disposed) {
console.error('[WebRTC] PeerConnection was disposed after createAnswer');
throw new Error('PeerConnection was disposed during answer creation');
}
// Diagnostic: log video m-line from answer SDP
if (answer.sdp) {
const videoMLine = answer.sdp.split('\n').find((l: string) => l.startsWith('m=video'));
const videoIdx = answer.sdp.indexOf('m=video');
const videoDir = videoIdx >= 0
? answer.sdp.split('\n').find((l: string) => l.startsWith('a=') && answer.sdp.indexOf(l) > videoIdx && (l.includes('sendrecv') || l.includes('recvonly') || l.includes('sendonly') || l.includes('inactive')))
: null;
console.log('[WebRTC] Answer video m-line:', videoMLine, 'video direction:', videoDir);
}
await this.peerConnection.setLocalDescription(answer);
console.log('[WebRTC] Local description set successfully');
// Process pending candidates after local description is set
await this.processPendingCandidates();
return answer;
@@ -310,23 +344,60 @@ class WebRTCManager {
if (!description.sdp) {
throw new Error('setRemoteDescription: sdp is required');
}
console.log('[WebRTC] Setting remote description, type:', description.type);
// When receiving an offer, ensure video transceiver direction is compatible
// react-native-webrtc may not auto-negotiate transceiver direction from remote SDP
if (description.type === 'offer' && description.sdp) {
this.ensureVideoTransceiverRecvForOffer(description.sdp);
}
const desc = new RTCSessionDescription({
type: description.type,
sdp: description.sdp,
});
await this.peerConnection.setRemoteDescription(desc);
console.log('[WebRTC] Remote description set successfully');
// Process pending candidates after remote description is set
await this.processPendingCandidates();
console.log('[WebRTC] Pending candidates processed, connection state:', this.peerConnection?.signalingState);
}
/**
* When receiving an offer where remote wants to send video,
* ensure our video transceiver direction allows receiving (recvonly).
* react-native-webrtc doesn't always auto-negotiate direction from remote SDP.
*/
private ensureVideoTransceiverRecvForOffer(remoteSdp: string): void {
if (!this.peerConnection) return;
// Check if the SDP contains a video media line with send direction
// react-native-webrtc may use \n instead of \r\n
const hasVideoSend =
remoteSdp.includes('m=video') &&
(remoteSdp.includes('a=sendrecv') || remoteSdp.includes('a=sendonly'));
if (!hasVideoSend) {
console.log('[WebRTC] Remote offer does not send video');
return;
}
const transceivers = this.peerConnection.getTransceivers();
const videoTransceiver = transceivers.find(t => t.receiver.track?.kind === 'video');
if (videoTransceiver) {
// If our direction is inactive, update to recvonly so we can receive video
if (videoTransceiver.direction === 'inactive') {
console.log('[WebRTC] Updating video transceiver from inactive to recvonly for remote video');
videoTransceiver.direction = 'recvonly';
}
}
}
/**
* Rollback to stable state (for Glare handling)
*/
@@ -397,8 +468,8 @@ class WebRTCManager {
}
/**
* Enable video using transceiver direction
* This preserves m-line order and triggers renegotiation
* Enable video using addTrack for maximum compatibility
* Actively triggers renegotiation
*/
async enableVideo(): Promise<MediaStream> {
if (!this.peerConnection) throw new Error('PeerConnection not initialized');
@@ -414,43 +485,53 @@ class WebRTCManager {
const videoTrack = videoStream.getVideoTracks()[0];
// Find the video transceiver and update it
// Find the video transceiver and replace track on it
const transceivers = this.peerConnection.getTransceivers();
const videoTransceiver = transceivers.find(t => t.receiver.track?.kind === 'video');
if (videoTransceiver) {
// Replace the track
// Replace the track on existing sender
await videoTransceiver.sender.replaceTrack(videoTrack);
// Update direction to sendrecv
// Force direction to sendrecv
videoTransceiver.direction = 'sendrecv';
console.log('[WebRTC] Video transceiver updated, direction:', videoTransceiver.direction);
} else {
console.error('[WebRTC] No video transceiver found!');
// No transceiver exists, add track directly (will create one)
this.peerConnection.addTrack(videoTrack);
console.log('[WebRTC] Video track added via addTrack (no existing transceiver)');
}
// Update local stream
const newStream = new MediaStream();
if (this.localStream) {
// Add existing audio tracks
this.localStream.getAudioTracks().forEach(track => {
newStream.addTrack(track);
});
// Stop old video tracks
this.localStream.getVideoTracks().forEach(track => {
track.stop();
});
}
// Add new video track
newStream.addTrack(videoTrack);
this.localStream = newStream;
console.log('[WebRTC] Video enabled successfully');
// onnegotiationneeded will be triggered automatically
console.log('[WebRTC] Video enabled, creating renegotiation offer...');
// Set lock to prevent onnegotiationneeded from firing duplicate
this.isNegotiating = true;
const offer = await this.createOffer();
if (this.peerConnection && !this.disposed) {
this.emit({ type: 'negotiationneeded', offer });
}
// Release lock after a delay
setTimeout(() => {
this.isNegotiating = false;
}, 500);
return newStream;
} catch (error) {
this.isNegotiating = false;
console.error('[WebRTC] Failed to enable video:', error);
throw error;
}
@@ -458,6 +539,7 @@ class WebRTCManager {
/**
* Disable video using transceiver direction
* Actively triggers renegotiation instead of relying on onnegotiationneeded
*/
async disableVideo(): Promise<MediaStream | null> {
if (!this.peerConnection) {
@@ -494,11 +576,25 @@ class WebRTCManager {
this.localStream.getAudioTracks().forEach(track => {
newStream.addTrack(track);
});
this.localStream = newStream;
console.log('[WebRTC] Video disabled successfully');
// onnegotiationneeded will be triggered automatically
this.localStream = newStream;
console.log('[WebRTC] Video disabled, creating renegotiation offer...');
// Set lock to prevent onnegotiationneeded from firing duplicate
this.isNegotiating = true;
try {
const offer = await this.createOffer();
if (this.peerConnection && !this.disposed) {
this.emit({ type: 'negotiationneeded', offer });
}
} catch (err) {
console.error('[WebRTC] Failed to create renegotiation offer for disableVideo:', err);
} finally {
setTimeout(() => {
this.isNegotiating = false;
}, 500);
}
return newStream;
}
@@ -559,7 +655,6 @@ class WebRTCManager {
if (this.remoteStream) {
this.remoteStream.getTracks().forEach((track) => track.stop());
this.remoteStream.release();
this.remoteStream = null;
}

View File

@@ -202,7 +202,7 @@ async function handleIncomingOffer(msg: WSCallSDPMessage, myUserId: string): Pro
return;
}
const signalingState = pc.signalingState;
let signalingState = pc.signalingState;
console.log('[CallStore] Received offer, signalingState:', signalingState);
// Glare handling: if we're not in stable state, we have a conflict
@@ -223,6 +223,16 @@ async function handleIncomingOffer(msg: WSCallSDPMessage, myUserId: string): Pro
await webrtcManager.rollback();
} catch (err) {
console.error('[CallStore] Rollback failed:', err);
// If rollback fails because we're already stable, that's fine - just proceed
if (pc.signalingState !== 'stable') {
return;
}
}
// Re-check state after rollback
signalingState = pc.signalingState;
if (signalingState !== 'stable') {
console.warn('[CallStore] Not stable after rollback, state:', signalingState);
return;
}
}