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✨ feat: stream back audio to user (test)

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Marco Beretta 2025-01-03 23:14:18 +01:00 • committed by Danny Avila
parent 964d47cfa3
commit 7aed891838
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6 changed files with 151 additions and 49 deletions
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60
client/src/hooks/useCall.ts
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@ -54,6 +54,46 @@ const useCall = () => {
};
}, []);
const handleRemoteStream = (stream: MediaStream | null) => {
console.log('[WebRTC] Remote stream received:', {
stream: stream,
active: stream?.active,
tracks: stream?.getTracks().map((t) => ({
kind: t.kind,
enabled: t.enabled,
muted: t.muted,
})),
});
if (!stream) {
console.error('[WebRTC] Received null remote stream');
updateStatus({
error: {
code: 'NO_REMOTE_STREAM',
message: 'No remote stream received',
},
});
return;
}
const audioTracks = stream.getAudioTracks();
if (!audioTracks.length) {
console.error('[WebRTC] No audio tracks in remote stream');
updateStatus({
error: {
code: 'NO_AUDIO_TRACKS',
message: 'Remote stream contains no audio',
},
});
return;
}
updateStatus({
remoteStream: stream,
callState: CallState.ACTIVE,
});
};
const handleConnectionStateChange = useCallback(
(state: ConnectionState) => {
switch (state) {
@ -123,6 +163,7 @@ const useCall = () => {
const startCall = useCallback(async () => {
if (!isConnected) {
console.log('Cannot start call - not connected to server');
updateStatus({
callState: CallState.ERROR,
error: {
@ -134,7 +175,10 @@ const useCall = () => {
}
try {
console.log('Starting new call...');
if (webrtcServiceRef.current) {
console.log('Cleaning up existing WebRTC connection');
webrtcServiceRef.current.close();
}
@ -149,13 +193,15 @@ const useCall = () => {
debug: true,
});
webrtcServiceRef.current.on('connectionStateChange', handleConnectionStateChange);
webrtcServiceRef.current.on('remoteStream', (stream: MediaStream) => {
updateStatus({ remoteStream: stream });
webrtcServiceRef.current.on('connectionStateChange', (state: ConnectionState) => {
console.log('WebRTC connection state changed:', state);
handleConnectionStateChange(state);
});
webrtcServiceRef.current.on('remoteStream', handleRemoteStream);
webrtcServiceRef.current.on('error', (error: string) => {
console.error('WebRTC error:', error);
updateStatus({
callState: CallState.ERROR,
isConnecting: false,
@ -166,9 +212,13 @@ const useCall = () => {
});
});
console.log('Initializing WebRTC connection...');
await webrtcServiceRef.current.initialize();
console.log('WebRTC initialization complete');
startConnectionMonitoring();
} catch (error) {
console.error('Failed to start call:', error);
updateStatus({
callState: CallState.ERROR,
isConnecting: false,
@ -203,9 +253,11 @@ const useCall = () => {
useEffect(() => {
const cleanupFns = [
addEventListener(WebSocketEvents.WEBRTC_ANSWER, (answer: RTCSessionDescriptionInit) => {
console.log('Received WebRTC answer:', answer);
webrtcServiceRef.current?.handleAnswer(answer);
}),
addEventListener(WebSocketEvents.ICE_CANDIDATE, (candidate: RTCIceCandidateInit) => {
console.log('Received ICE candidate:', candidate);
webrtcServiceRef.current?.addIceCandidate(candidate);
}),
];