Merge pull request #228 from kerberos-io/feature/improve-webrtc-tracing

feature/improve-webrtc-tracing

.github/workflows/release-create.yml

@@ -44,11 +44,6 @@ jobs:
         run: |
           docker tag ${{matrix.architecture}} $REPO-arch:arch-${{matrix.architecture}}-${{github.event.inputs.tag || github.ref_name}}
           docker push $REPO-arch:arch-${{matrix.architecture}}-${{github.event.inputs.tag || github.ref_name}}
-      - name: Create new manifest
-        run: docker manifest create $REPO:${{ github.event.inputs.tag || github.ref_name }} $REPO-arch:arch-${{matrix.architecture}}-${{github.event.inputs.tag || github.ref_name}}
-      - name: Create latest manifest
-        run: docker manifest create $REPO:latest $REPO-arch:arch-${{matrix.architecture}}-${{github.event.inputs.tag || github.ref_name}}
-        if: github.event.inputs.tag == 'test'
       - name: Upload artifact
         uses: actions/upload-artifact@v4
         with:

machinery/main.go

@@ -101,31 +101,35 @@ func main() {
 	switch action {
 
 	case "version":
-		log.Log.Info("main.Main(): You are currrently running Kerberos Agent " + VERSION)
-
+		{
+			log.Log.Info("main.Main(): You are currrently running Kerberos Agent " + VERSION)
+		}
 	case "discover":
-		// Convert duration to int
-		timeout, err := time.ParseDuration(timeout + "ms")
-		if err != nil {
-			log.Log.Fatal("main.Main(): could not parse timeout: " + err.Error())
-			return
+		{
+			// Convert duration to int
+			timeout, err := time.ParseDuration(timeout + "ms")
+			if err != nil {
+				log.Log.Fatal("main.Main(): could not parse timeout: " + err.Error())
+				return
+			}
+			onvif.Discover(timeout)
 		}
-		onvif.Discover(timeout)
-
 	case "decrypt":
-		log.Log.Info("main.Main(): Decrypting: " + flag.Arg(0) + " with key: " + flag.Arg(1))
-		symmetricKey := []byte(flag.Arg(1))
+		{
+			log.Log.Info("main.Main(): Decrypting: " + flag.Arg(0) + " with key: " + flag.Arg(1))
+			symmetricKey := []byte(flag.Arg(1))
 
-		if symmetricKey == nil || len(symmetricKey) == 0 {
-			log.Log.Fatal("main.Main(): symmetric key should not be empty")
-			return
-		}
-		if len(symmetricKey) != 32 {
-			log.Log.Fatal("main.Main(): symmetric key should be 32 bytes")
-			return
-		}
+			if len(symmetricKey) == 0 {
+				log.Log.Fatal("main.Main(): symmetric key should not be empty")
+				return
+			}
+			if len(symmetricKey) != 32 {
+				log.Log.Fatal("main.Main(): symmetric key should be 32 bytes")
+				return
+			}
 
-		utils.Decrypt(flag.Arg(0), symmetricKey)
+			utils.Decrypt(flag.Arg(0), symmetricKey)
+		}
 
 	case "run":
 		{
@@ -213,6 +217,8 @@ func main() {
 			routers.StartWebserver(configDirectory, &configuration, &communication, &capture)
 		}
 	default:
-		log.Log.Error("main.Main(): Sorry I don't understand :(")
+		{
+			log.Log.Error("main.Main(): Sorry I don't understand :(")
+		}
 	}
 }

machinery/src/cloud/Cloud.go

@@ -804,6 +804,12 @@ func HandleLiveStreamHD(livestreamCursor *packets.QueueCursor, configuration *mo
 			streams, _ := rtspClient.GetStreams()
 			videoTrack := webrtc.NewVideoTrack(streams)
 			audioTrack := webrtc.NewAudioTrack(streams)
+
+			if videoTrack == nil && audioTrack == nil {
+				log.Log.Error("cloud.HandleLiveStreamHD(): failed to create both video and audio tracks")
+				return
+			}
+
 			go webrtc.WriteToTrack(livestreamCursor, configuration, communication, mqttClient, videoTrack, audioTrack, rtspClient)
 
 			if config.Capture.ForwardWebRTC == "true" {

machinery/src/routers/mqtt/main.go

@@ -123,7 +123,6 @@ func ConfigureMQTT(configDirectory string, configuration *models.Configuration,
 			opts.SetClientID(mqttClientID)
 			log.Log.Info("routers.mqtt.main.ConfigureMQTT(): Set ClientID " + mqttClientID)
 			rand.Seed(time.Now().UnixNano())
-			webrtc.CandidateArrays = make(map[string](chan string))
 
 			opts.OnConnect = func(c mqtt.Client) {
 				// We managed to connect to the MQTT broker, hurray!

machinery/src/webrtc/main.go

@@ -1,6 +1,7 @@
 package webrtc
 
 import (
+	"context"
 	"encoding/base64"
 	"encoding/json"
 	"io"
@@ -22,13 +23,104 @@ import (
 	pionMedia "github.com/pion/webrtc/v4/pkg/media"
 )
 
-var (
-	CandidatesMutex     sync.Mutex
-	CandidateArrays     map[string](chan string)
-	peerConnectionCount int64
-	peerConnections     map[string]*pionWebRTC.PeerConnection
+const (
+	// Channel buffer sizes
+	candidateChannelBuffer = 100
+	rtcpBufferSize         = 1500
+
+	// Timeouts and intervals
+	keepAliveTimeout = 15 * time.Second
+	defaultTimeout   = 10 * time.Second
+
+	// Track identifiers
+	trackStreamID = "kerberos-stream"
 )
 
+// ConnectionManager manages WebRTC peer connections in a thread-safe manner
+type ConnectionManager struct {
+	mu                  sync.RWMutex
+	candidateChannels   map[string]chan string
+	peerConnections     map[string]*peerConnectionWrapper
+	peerConnectionCount int64
+}
+
+// peerConnectionWrapper wraps a peer connection with additional metadata
+type peerConnectionWrapper struct {
+	conn      *pionWebRTC.PeerConnection
+	cancelCtx context.CancelFunc
+	done      chan struct{}
+}
+
+var globalConnectionManager = NewConnectionManager()
+
+// NewConnectionManager creates a new connection manager
+func NewConnectionManager() *ConnectionManager {
+	return &ConnectionManager{
+		candidateChannels: make(map[string]chan string),
+		peerConnections:   make(map[string]*peerConnectionWrapper),
+	}
+}
+
+// GetOrCreateCandidateChannel gets or creates a candidate channel for a session
+func (cm *ConnectionManager) GetOrCreateCandidateChannel(sessionKey string) chan string {
+	cm.mu.Lock()
+	defer cm.mu.Unlock()
+
+	if ch, exists := cm.candidateChannels[sessionKey]; exists {
+		return ch
+	}
+
+	ch := make(chan string, candidateChannelBuffer)
+	cm.candidateChannels[sessionKey] = ch
+	return ch
+}
+
+// CloseCandidateChannel safely closes and removes a candidate channel
+func (cm *ConnectionManager) CloseCandidateChannel(sessionKey string) {
+	cm.mu.Lock()
+	defer cm.mu.Unlock()
+
+	if ch, exists := cm.candidateChannels[sessionKey]; exists {
+		close(ch)
+		delete(cm.candidateChannels, sessionKey)
+	}
+}
+
+// AddPeerConnection adds a peer connection to the manager
+func (cm *ConnectionManager) AddPeerConnection(sessionID string, wrapper *peerConnectionWrapper) {
+	cm.mu.Lock()
+	defer cm.mu.Unlock()
+	cm.peerConnections[sessionID] = wrapper
+}
+
+// RemovePeerConnection removes a peer connection from the manager
+func (cm *ConnectionManager) RemovePeerConnection(sessionID string) {
+	cm.mu.Lock()
+	defer cm.mu.Unlock()
+
+	if wrapper, exists := cm.peerConnections[sessionID]; exists {
+		if wrapper.cancelCtx != nil {
+			wrapper.cancelCtx()
+		}
+		delete(cm.peerConnections, sessionID)
+	}
+}
+
+// GetPeerConnectionCount returns the current count of active peer connections
+func (cm *ConnectionManager) GetPeerConnectionCount() int64 {
+	return atomic.LoadInt64(&cm.peerConnectionCount)
+}
+
+// IncrementPeerCount atomically increments the peer connection count
+func (cm *ConnectionManager) IncrementPeerCount() int64 {
+	return atomic.AddInt64(&cm.peerConnectionCount, 1)
+}
+
+// DecrementPeerCount atomically decrements the peer connection count
+func (cm *ConnectionManager) DecrementPeerCount() int64 {
+	return atomic.AddInt64(&cm.peerConnectionCount, -1)
+}
+
 type WebRTC struct {
 	Name                  string
 	StunServers           []string
@@ -46,7 +138,7 @@ func CreateWebRTC(name string, stunServers []string, turnServers []string, turnS
 		TurnServers:           turnServers,
 		TurnServersUsername:   turnServersUsername,
 		TurnServersCredential: turnServersCredential,
-		Timer:                 time.NewTimer(time.Second * 10),
+		Timer:                 time.NewTimer(defaultTimeout),
 	}
 }
 
@@ -68,19 +160,14 @@ func (w WebRTC) CreateOffer(sd []byte) pionWebRTC.SessionDescription {
 }
 
 func RegisterCandidates(key string, candidate models.ReceiveHDCandidatesPayload) {
-	// Set lock
-	CandidatesMutex.Lock()
-	_, ok := CandidateArrays[key]
-	if !ok {
-		CandidateArrays[key] = make(chan string, 100)
-	}
-	log.Log.Info("webrtc.main.HandleReceiveHDCandidates(): " + candidate.Candidate)
+	ch := globalConnectionManager.GetOrCreateCandidateChannel(key)
+
+	log.Log.Info("webrtc.main.RegisterCandidates(): " + candidate.Candidate)
 	select {
-	case CandidateArrays[key] <- candidate.Candidate:
+	case ch <- candidate.Candidate:
 	default:
-		log.Log.Info("webrtc.main.HandleReceiveHDCandidates(): channel is full.")
+		log.Log.Info("webrtc.main.RegisterCandidates(): channel is full, dropping candidate")
 	}
-	CandidatesMutex.Unlock()
 }
 
 func RegisterDefaultInterceptors(mediaEngine *pionWebRTC.MediaEngine, interceptorRegistry *interceptor.Registry) error {
@@ -107,12 +194,7 @@ func InitializeWebRTCConnection(configuration *models.Configuration, communicati
 
 	// We create a channel which will hold the candidates for this session.
 	sessionKey := config.Key + "/" + handshake.SessionID
-	CandidatesMutex.Lock()
-	_, ok := CandidateArrays[sessionKey]
-	if !ok {
-		CandidateArrays[sessionKey] = make(chan string, 100)
-	}
-	CandidatesMutex.Unlock()
+	candidateChannel := globalConnectionManager.GetOrCreateCandidateChannel(sessionKey)
 
 	// Set variables
 	hubKey := handshake.HubKey
@@ -178,81 +260,126 @@ func InitializeWebRTCConnection(configuration *models.Configuration, communicati
 
 		if err == nil && peerConnection != nil {
 
-			var videoSender *pionWebRTC.RTPSender = nil
-			if videoSender, err = peerConnection.AddTrack(videoTrack); err != nil {
-				log.Log.Error("webrtc.main.InitializeWebRTCConnection(): something went wrong while adding video track: " + err.Error())
+			// Create context for this connection
+			ctx, cancel := context.WithCancel(context.Background())
+			wrapper := &peerConnectionWrapper{
+				conn:      peerConnection,
+				cancelCtx: cancel,
+				done:      make(chan struct{}),
 			}
+
+			var videoSender *pionWebRTC.RTPSender = nil
+			if videoTrack != nil {
+				if videoSender, err = peerConnection.AddTrack(videoTrack); err != nil {
+					log.Log.Error("webrtc.main.InitializeWebRTCConnection(): error adding video track: " + err.Error())
+					cancel()
+					return
+				}
+			} else {
+				log.Log.Info("webrtc.main.InitializeWebRTCConnection(): video track is nil, skipping video")
+			}
+
 			// Read incoming RTCP packets
 			// Before these packets are returned they are processed by interceptors. For things
 			// like NACK this needs to be called.
-			go func() {
-				rtcpBuf := make([]byte, 1500)
-				for {
-					if _, _, rtcpErr := videoSender.Read(rtcpBuf); rtcpErr != nil {
-						return
+			if videoSender != nil {
+				go func() {
+					defer func() {
+						log.Log.Info("webrtc.main.InitializeWebRTCConnection(): video RTCP reader stopped")
+					}()
+					rtcpBuf := make([]byte, rtcpBufferSize)
+					for {
+						select {
+						case <-ctx.Done():
+							return
+						default:
+							if _, _, rtcpErr := videoSender.Read(rtcpBuf); rtcpErr != nil {
+								return
+							}
+						}
 					}
-				}
-			}()
+				}()
+			}
 
 			var audioSender *pionWebRTC.RTPSender = nil
-			if audioSender, err = peerConnection.AddTrack(audioTrack); err != nil {
-				log.Log.Error("webrtc.main.InitializeWebRTCConnection(): something went wrong while adding audio track: " + err.Error())
-			} // Read incoming RTCP packets
+			if audioTrack != nil {
+				if audioSender, err = peerConnection.AddTrack(audioTrack); err != nil {
+					log.Log.Error("webrtc.main.InitializeWebRTCConnection(): error adding audio track: " + err.Error())
+					cancel()
+					return
+				}
+			} else {
+				log.Log.Info("webrtc.main.InitializeWebRTCConnection(): audio track is nil, skipping audio")
+			}
+
+			// Read incoming RTCP packets
 			// Before these packets are returned they are processed by interceptors. For things
 			// like NACK this needs to be called.
-			go func() {
-				rtcpBuf := make([]byte, 1500)
-				for {
-					if _, _, rtcpErr := audioSender.Read(rtcpBuf); rtcpErr != nil {
-						return
+			if audioSender != nil {
+				go func() {
+					defer func() {
+						log.Log.Info("webrtc.main.InitializeWebRTCConnection(): audio RTCP reader stopped")
+					}()
+					rtcpBuf := make([]byte, rtcpBufferSize)
+					for {
+						select {
+						case <-ctx.Done():
+							return
+						default:
+							if _, _, rtcpErr := audioSender.Read(rtcpBuf); rtcpErr != nil {
+								return
+							}
+						}
 					}
-				}
-			}()
+				}()
+			}
 
 			peerConnection.OnConnectionStateChange(func(connectionState pionWebRTC.PeerConnectionState) {
-				if connectionState == pionWebRTC.PeerConnectionStateDisconnected || connectionState == pionWebRTC.PeerConnectionStateClosed {
-					// Set lock
-					CandidatesMutex.Lock()
-					atomic.AddInt64(&peerConnectionCount, -1)
-					_, ok := CandidateArrays[sessionKey]
-					if ok {
-						close(CandidateArrays[sessionKey])
-						delete(CandidateArrays, sessionKey)
-					}
-					// Not really needed.
-					//senders := peerConnection.GetSenders()
-					//for _, sender := range senders {
-					//	if err := peerConnection.RemoveTrack(sender); err != nil {
-					//		log.Log.Error("webrtc.main.InitializeWebRTCConnection(): something went wrong while removing track: " + err.Error())
-					//	}
-					//}
-					if err := peerConnection.Close(); err != nil {
-						log.Log.Error("webrtc.main.InitializeWebRTCConnection(): something went wrong while closing peer connection: " + err.Error())
-					}
-					peerConnections[handshake.SessionID] = nil
-					delete(peerConnections, handshake.SessionID)
-					CandidatesMutex.Unlock()
-				} else if connectionState == pionWebRTC.PeerConnectionStateConnected {
-					CandidatesMutex.Lock()
-					atomic.AddInt64(&peerConnectionCount, 1)
-					CandidatesMutex.Unlock()
-				} else if connectionState == pionWebRTC.PeerConnectionStateFailed {
-					log.Log.Info("webrtc.main.InitializeWebRTCConnection(): ICEConnectionStateFailed")
-				}
 				log.Log.Info("webrtc.main.InitializeWebRTCConnection(): connection state changed to: " + connectionState.String())
-				log.Log.Info("webrtc.main.InitializeWebRTCConnection(): Number of peers connected (" + strconv.FormatInt(peerConnectionCount, 10) + ")")
+
+				switch connectionState {
+				case pionWebRTC.PeerConnectionStateDisconnected, pionWebRTC.PeerConnectionStateClosed:
+					count := globalConnectionManager.DecrementPeerCount()
+					log.Log.Info("webrtc.main.InitializeWebRTCConnection(): Peer disconnected. Active peers: " + string(rune(count)))
+
+					// Clean up resources
+					globalConnectionManager.CloseCandidateChannel(sessionKey)
+
+					if err := peerConnection.Close(); err != nil {
+						log.Log.Error("webrtc.main.InitializeWebRTCConnection(): error closing peer connection: " + err.Error())
+					}
+
+					globalConnectionManager.RemovePeerConnection(handshake.SessionID)
+					close(wrapper.done)
+
+				case pionWebRTC.PeerConnectionStateConnected:
+					count := globalConnectionManager.IncrementPeerCount()
+					log.Log.Info("webrtc.main.InitializeWebRTCConnection(): Peer connected. Active peers: " + string(rune(count)))
+
+				case pionWebRTC.PeerConnectionStateFailed:
+					log.Log.Info("webrtc.main.InitializeWebRTCConnection(): ICE connection failed")
+				}
 			})
 
 			go func() {
+				defer func() {
+					log.Log.Info("webrtc.main.InitializeWebRTCConnection(): candidate processor stopped for session: " + handshake.SessionID)
+				}()
+
 				// Iterate over the candidates and send them to the remote client
-				// Non blocking channe
-				for candidate := range CandidateArrays[sessionKey] {
-					CandidatesMutex.Lock()
-					log.Log.Info(">>>> webrtc.main.InitializeWebRTCConnection(): Received candidate from channel: " + candidate)
-					if candidateErr := peerConnection.AddICECandidate(pionWebRTC.ICECandidateInit{Candidate: string(candidate)}); candidateErr != nil {
-						log.Log.Error("webrtc.main.InitializeWebRTCConnection(): something went wrong while adding candidate: " + candidateErr.Error())
+				for {
+					select {
+					case <-ctx.Done():
+						return
+					case candidate, ok := <-candidateChannel:
+						if !ok {
+							return
+						}
+						log.Log.Info("webrtc.main.InitializeWebRTCConnection(): Received candidate from channel: " + candidate)
+						if candidateErr := peerConnection.AddICECandidate(pionWebRTC.ICECandidateInit{Candidate: candidate}); candidateErr != nil {
+							log.Log.Error("webrtc.main.InitializeWebRTCConnection(): error adding candidate: " + candidateErr.Error())
+						}
 					}
-					CandidatesMutex.Unlock()
 				}
 			}()
 
@@ -270,22 +397,56 @@ func InitializeWebRTCConnection(configuration *models.Configuration, communicati
 
 			// When an ICE candidate is available send to the other peer using the signaling server (MQTT).
 			// The other peer will add this candidate by calling AddICECandidate
+			var hasRelayCandidates bool
 			peerConnection.OnICECandidate(func(candidate *pionWebRTC.ICECandidate) {
+
 				if candidate == nil {
+					log.Log.Info("webrtc.main.InitializeWebRTCConnection(): ICE gathering complete (candidate is nil)")
+					if !hasRelayCandidates {
+						log.Log.Error("webrtc.main.InitializeWebRTCConnection(): WARNING - No TURN (relay) candidates were gathered! TURN servers: " +
+							config.TURNURI + ", Username: " + config.TURNUsername + ", ForceTurn: " + config.ForceTurn)
+					}
 					return
 				}
 
+				// Log candidate details for debugging
+				candidateJSON := candidate.ToJSON()
+				candidateStr := candidateJSON.Candidate
+
+				// Determine candidate type from the candidate string
+				candidateType := "unknown"
+				if candidateJSON.Candidate != "" {
+					switch candidate.Typ {
+					case pionWebRTC.ICECandidateTypeRelay:
+						candidateType = "relay"
+					case pionWebRTC.ICECandidateTypeSrflx:
+						candidateType = "srflx"
+					case pionWebRTC.ICECandidateTypeHost:
+						candidateType = "host"
+					case pionWebRTC.ICECandidateTypePrflx:
+						candidateType = "prflx"
+					}
+				}
+
+				// Track if we received any relay (TURN) candidates
+				if candidateType == "relay" {
+					hasRelayCandidates = true
+				}
+
+				log.Log.Info("webrtc.main.InitializeWebRTCConnection(): ICE candidate received - Type: " + candidateType +
+					", Candidate: " + candidateStr)
+
 				//  Create a config map
 				valueMap := make(map[string]interface{})
-				candateJSON := candidate.ToJSON()
-				candateBinary, err := json.Marshal(candateJSON)
+				candateBinary, err := json.Marshal(candidateJSON)
 				if err == nil {
 					valueMap["candidate"] = string(candateBinary)
 					// SDP is not needed to be send..
 					//valueMap["sdp"] = []byte(base64.StdEncoding.EncodeToString([]byte(answer.SDP)))
 					valueMap["session_id"] = handshake.SessionID
+					log.Log.Info("webrtc.main.InitializeWebRTCConnection(): sending " + candidateType + " candidate to hub")
 				} else {
-					log.Log.Info("webrtc.main.InitializeWebRTCConnection(): something went wrong while marshalling candidate: " + err.Error())
+					log.Log.Error("webrtc.main.InitializeWebRTCConnection(): failed to marshal candidate: " + err.Error())
 				}
 
 				// We'll send the candidate to the hub
@@ -305,8 +466,8 @@ func InitializeWebRTCConnection(configuration *models.Configuration, communicati
 				}
 			})
 
-			// Create a channel which will be used to send candidates to the other peer
-			peerConnections[handshake.SessionID] = peerConnection
+			// Store peer connection in manager
+			globalConnectionManager.AddPeerConnection(handshake.SessionID, wrapper)
 
 			if err == nil {
 				//  Create a config map
@@ -339,7 +500,11 @@ func InitializeWebRTCConnection(configuration *models.Configuration, communicati
 
 func NewVideoTrack(streams []packets.Stream) *pionWebRTC.TrackLocalStaticSample {
 	mimeType := pionWebRTC.MimeTypeH264
-	outboundVideoTrack, _ := pionWebRTC.NewTrackLocalStaticSample(pionWebRTC.RTPCodecCapability{MimeType: mimeType}, "video", "pion124")
+	outboundVideoTrack, err := pionWebRTC.NewTrackLocalStaticSample(pionWebRTC.RTPCodecCapability{MimeType: mimeType}, "video", trackStreamID)
+	if err != nil {
+		log.Log.Error("webrtc.main.NewVideoTrack(): error creating video track: " + err.Error())
+		return nil
+	}
 	return outboundVideoTrack
 }
 
@@ -354,161 +519,245 @@ func NewAudioTrack(streams []packets.Stream) *pionWebRTC.TrackLocalStaticSample
 			mimeType = pionWebRTC.MimeTypePCMA
 		}
 	}
-	outboundAudioTrack, _ := pionWebRTC.NewTrackLocalStaticSample(pionWebRTC.RTPCodecCapability{MimeType: mimeType}, "audio", "pion124")
+	if mimeType == "" {
+		log.Log.Error("webrtc.main.NewAudioTrack(): no supported audio codec found")
+		return nil
+	}
+	outboundAudioTrack, err := pionWebRTC.NewTrackLocalStaticSample(pionWebRTC.RTPCodecCapability{MimeType: mimeType}, "audio", trackStreamID)
+	if err != nil {
+		log.Log.Error("webrtc.main.NewAudioTrack(): error creating audio track: " + err.Error())
+		return nil
+	}
 	return outboundAudioTrack
 }
 
+// streamState holds state information for the streaming process
+type streamState struct {
+	lastKeepAlive    int64
+	peerCount        int64
+	start            bool
+	receivedKeyFrame bool
+	lastAudioSample  *pionMedia.Sample
+	lastVideoSample  *pionMedia.Sample
+}
+
+// codecSupport tracks which codecs are available in the stream
+type codecSupport struct {
+	hasH264      bool
+	hasPCM_MULAW bool
+	hasAAC       bool
+	hasOpus      bool
+}
+
+// detectCodecs examines the stream to determine which codecs are available
+func detectCodecs(rtspClient capture.RTSPClient) codecSupport {
+	support := codecSupport{}
+	streams, _ := rtspClient.GetStreams()
+
+	for _, stream := range streams {
+		switch stream.Name {
+		case "H264":
+			support.hasH264 = true
+		case "PCM_MULAW":
+			support.hasPCM_MULAW = true
+		case "AAC":
+			support.hasAAC = true
+		case "OPUS":
+			support.hasOpus = true
+		}
+	}
+
+	return support
+}
+
+// hasValidCodecs checks if at least one valid video or audio codec is present
+func (cs codecSupport) hasValidCodecs() bool {
+	hasVideo := cs.hasH264
+	hasAudio := cs.hasPCM_MULAW || cs.hasAAC || cs.hasOpus
+	return hasVideo || hasAudio
+}
+
+// shouldContinueStreaming determines if streaming should continue based on keepalive and peer count
+func shouldContinueStreaming(config models.Config, state *streamState) bool {
+	if config.Capture.ForwardWebRTC != "true" {
+		return true
+	}
+
+	now := time.Now().Unix()
+	hasTimedOut := (now - state.lastKeepAlive) > int64(keepAliveTimeout.Seconds())
+	hasNoPeers := state.peerCount == 0
+
+	return !hasTimedOut && !hasNoPeers
+}
+
+// updateStreamState updates keepalive and peer count from communication channels
+func updateStreamState(communication *models.Communication, state *streamState) {
+	select {
+	case keepAliveStr := <-communication.HandleLiveHDKeepalive:
+		if val, err := strconv.ParseInt(keepAliveStr, 10, 64); err == nil {
+			state.lastKeepAlive = val
+		}
+	default:
+	}
+
+	select {
+	case peerCountStr := <-communication.HandleLiveHDPeers:
+		if val, err := strconv.ParseInt(peerCountStr, 10, 64); err == nil {
+			state.peerCount = val
+		}
+	default:
+	}
+}
+
+// writeFinalSamples writes any remaining buffered samples
+func writeFinalSamples(state *streamState, videoTrack, audioTrack *pionWebRTC.TrackLocalStaticSample) {
+	if state.lastVideoSample != nil && videoTrack != nil {
+		if err := videoTrack.WriteSample(*state.lastVideoSample); err != nil && err != io.ErrClosedPipe {
+			log.Log.Error("webrtc.main.writeFinalSamples(): error writing final video sample: " + err.Error())
+		}
+	}
+
+	if state.lastAudioSample != nil && audioTrack != nil {
+		if err := audioTrack.WriteSample(*state.lastAudioSample); err != nil && err != io.ErrClosedPipe {
+			log.Log.Error("webrtc.main.writeFinalSamples(): error writing final audio sample: " + err.Error())
+		}
+	}
+}
+
+// processVideoPacket processes a video packet and writes samples to the track
+func processVideoPacket(pkt packets.Packet, state *streamState, videoTrack *pionWebRTC.TrackLocalStaticSample, config models.Config) {
+	if videoTrack == nil {
+		return
+	}
+
+	// Start at the first keyframe
+	if pkt.IsKeyFrame {
+		state.start = true
+	}
+
+	if !state.start {
+		return
+	}
+
+	sample := pionMedia.Sample{Data: pkt.Data, PacketTimestamp: uint32(pkt.Time)}
+
+	if config.Capture.ForwardWebRTC == "true" {
+		// Remote forwarding not yet implemented
+		log.Log.Debug("webrtc.main.processVideoPacket(): remote forwarding not implemented")
+		return
+	}
+
+	if state.lastVideoSample != nil {
+		duration := sample.PacketTimestamp - state.lastVideoSample.PacketTimestamp
+		state.lastVideoSample.Duration = time.Duration(duration) * time.Millisecond
+
+		if err := videoTrack.WriteSample(*state.lastVideoSample); err != nil && err != io.ErrClosedPipe {
+			log.Log.Error("webrtc.main.processVideoPacket(): error writing video sample: " + err.Error())
+		}
+	}
+
+	state.lastVideoSample = &sample
+}
+
+// processAudioPacket processes an audio packet and writes samples to the track
+func processAudioPacket(pkt packets.Packet, state *streamState, audioTrack *pionWebRTC.TrackLocalStaticSample, hasAAC bool) {
+	if audioTrack == nil {
+		return
+	}
+
+	if hasAAC {
+		// AAC transcoding not yet implemented
+		// TODO: Implement AAC to PCM_MULAW transcoding
+		return
+	}
+
+	sample := pionMedia.Sample{Data: pkt.Data, PacketTimestamp: uint32(pkt.Time)}
+
+	if state.lastAudioSample != nil {
+		duration := sample.PacketTimestamp - state.lastAudioSample.PacketTimestamp
+		state.lastAudioSample.Duration = time.Duration(duration) * time.Millisecond
+
+		if err := audioTrack.WriteSample(*state.lastAudioSample); err != nil && err != io.ErrClosedPipe {
+			log.Log.Error("webrtc.main.processAudioPacket(): error writing audio sample: " + err.Error())
+		}
+	}
+
+	state.lastAudioSample = &sample
+}
+
 func WriteToTrack(livestreamCursor *packets.QueueCursor, configuration *models.Configuration, communication *models.Communication, mqttClient mqtt.Client, videoTrack *pionWebRTC.TrackLocalStaticSample, audioTrack *pionWebRTC.TrackLocalStaticSample, rtspClient capture.RTSPClient) {
 
 	config := configuration.Config
 
-	// Make peerconnection map
-	peerConnections = make(map[string]*pionWebRTC.PeerConnection)
-
-	// Set the indexes for the video & audio streams
-	// Later when we read a packet we need to figure out which track to send it to.
-	hasH264 := false
-	hasPCM_MULAW := false
-	hasAAC := false
-	hasOpus := false
-	streams, _ := rtspClient.GetStreams()
-	for _, stream := range streams {
-		if stream.Name == "H264" {
-			hasH264 = true
-		} else if stream.Name == "PCM_MULAW" {
-			hasPCM_MULAW = true
-		} else if stream.Name == "AAC" {
-			hasAAC = true
-		} else if stream.Name == "OPUS" {
-			hasOpus = true
-		}
+	// Check if at least one track is available
+	if videoTrack == nil && audioTrack == nil {
+		log.Log.Error("webrtc.main.WriteToTrack(): both video and audio tracks are nil, cannot proceed")
+		return
 	}
 
-	if !hasH264 && !hasPCM_MULAW && !hasAAC && !hasOpus {
-		log.Log.Error("webrtc.main.WriteToTrack(): no valid video codec and audio codec found.")
-	} else {
-		if config.Capture.TranscodingWebRTC == "true" {
-			// Todo..
-		} else {
-			//log.Log.Info("webrtc.main.WriteToTrack(): not using a transcoder.")
-		}
+	// Detect available codecs
+	codecs := detectCodecs(rtspClient)
 
-		var cursorError error
-		var pkt packets.Packet
-		var lastAudioSample *pionMedia.Sample = nil
-		var lastVideoSample *pionMedia.Sample = nil
-
-		start := false
-		receivedKeyFrame := false
-		lastKeepAlive := "0"
-		peerCount := "0"
-
-		for cursorError == nil {
-
-			pkt, cursorError = livestreamCursor.ReadPacket()
-
-			//if config.Capture.ForwardWebRTC != "true" && peerConnectionCount == 0 {
-			//	start = false
-			//	receivedKeyFrame = false
-			//	continue
-			//}
-
-			select {
-			case lastKeepAlive = <-communication.HandleLiveHDKeepalive:
-			default:
-			}
-
-			select {
-			case peerCount = <-communication.HandleLiveHDPeers:
-			default:
-			}
-
-			now := time.Now().Unix()
-			lastKeepAliveN, _ := strconv.ParseInt(lastKeepAlive, 10, 64)
-			hasTimedOut := (now - lastKeepAliveN) > 15 // if longer then no response in 15 sec.
-			hasNoPeers := peerCount == "0"
-
-			if config.Capture.ForwardWebRTC == "true" && (hasTimedOut || hasNoPeers) {
-				start = false
-				receivedKeyFrame = false
-				continue
-			}
-
-			if len(pkt.Data) == 0 || pkt.Data == nil {
-				receivedKeyFrame = false
-				continue
-			}
-
-			if !receivedKeyFrame {
-				if pkt.IsKeyFrame {
-					receivedKeyFrame = true
-				} else {
-					continue
-				}
-			}
-
-			//if config.Capture.TranscodingWebRTC == "true" {
-			// We will transcode the video
-			// TODO..
-			//}
-
-			if pkt.IsVideo {
-
-				// Start at the first keyframe
-				if pkt.IsKeyFrame {
-					start = true
-				}
-				if start {
-					sample := pionMedia.Sample{Data: pkt.Data, PacketTimestamp: uint32(pkt.Time)}
-					//sample = pionMedia.Sample{Data: pkt.Data, Duration: time.Second}
-					if config.Capture.ForwardWebRTC == "true" {
-						// We will send the video to a remote peer
-						// TODO..
-					} else {
-						if lastVideoSample != nil {
-							duration := sample.PacketTimestamp - lastVideoSample.PacketTimestamp
-							bufferDurationCasted := time.Duration(duration) * time.Millisecond
-							lastVideoSample.Duration = bufferDurationCasted
-							if err := videoTrack.WriteSample(*lastVideoSample); err != nil && err != io.ErrClosedPipe {
-								log.Log.Error("webrtc.main.WriteToTrack(): something went wrong while writing sample: " + err.Error())
-							}
-						}
-
-						lastVideoSample = &sample
-					}
-				}
-			} else if pkt.IsAudio {
-
-				// @TODO: We need to check if the audio is PCM_MULAW or AAC
-				// If AAC we need to transcode it to PCM_MULAW
-				// If PCM_MULAW we can send it directly.
-
-				if hasAAC {
-					// We will transcode the audio from AAC to PCM_MULAW
-					// Not sure how to do this yet, but we need to use a decoder
-					// and then encode it to PCM_MULAW.
-					// TODO..
-					//d := fdkaac.NewAacDecoder()
-					continue
-				}
-
-				// We will send the audio
-				sample := pionMedia.Sample{Data: pkt.Data, PacketTimestamp: uint32(pkt.Time)}
-
-				if lastAudioSample != nil {
-					duration := sample.PacketTimestamp - lastAudioSample.PacketTimestamp
-					bufferDurationCasted := time.Duration(duration) * time.Millisecond
-					lastAudioSample.Duration = bufferDurationCasted
-					if err := audioTrack.WriteSample(*lastAudioSample); err != nil && err != io.ErrClosedPipe {
-						log.Log.Error("webrtc.main.WriteToTrack(): something went wrong while writing sample: " + err.Error())
-					}
-				}
-
-				lastAudioSample = &sample
-			}
-		}
+	if !codecs.hasValidCodecs() {
+		log.Log.Error("webrtc.main.WriteToTrack(): no valid video or audio codec found")
+		return
 	}
 
-	peerConnectionCount = 0
-	log.Log.Info("webrtc.main.WriteToTrack(): stop writing to track.")
+	if config.Capture.TranscodingWebRTC == "true" {
+		log.Log.Info("webrtc.main.WriteToTrack(): transcoding enabled but not yet implemented")
+	}
+
+	// Initialize streaming state
+	state := &streamState{
+		lastKeepAlive: time.Now().Unix(),
+		peerCount:     0,
+	}
+
+	defer func() {
+		writeFinalSamples(state, videoTrack, audioTrack)
+		log.Log.Info("webrtc.main.WriteToTrack(): stopped writing to track")
+	}()
+
+	var pkt packets.Packet
+	var cursorError error
+
+	for cursorError == nil {
+		pkt, cursorError = livestreamCursor.ReadPacket()
+
+		if cursorError != nil {
+			break
+		}
+
+		// Update state from communication channels
+		updateStreamState(communication, state)
+
+		// Check if we should continue streaming
+		if !shouldContinueStreaming(config, state) {
+			state.start = false
+			state.receivedKeyFrame = false
+			continue
+		}
+
+		// Skip empty packets
+		if len(pkt.Data) == 0 || pkt.Data == nil {
+			state.receivedKeyFrame = false
+			continue
+		}
+
+		// Wait for first keyframe before processing
+		if !state.receivedKeyFrame {
+			if pkt.IsKeyFrame {
+				state.receivedKeyFrame = true
+			} else {
+				continue
+			}
+		}
+
+		// Process video or audio packets
+		if pkt.IsVideo {
+			processVideoPacket(pkt, state, videoTrack, config)
+		} else if pkt.IsAudio {
+			processAudioPacket(pkt, state, audioTrack, codecs.hasAAC)
+		}
+	}
 }