Update Darwin build job to use macOS 13 for consistency

.github/workflows/docker.yml

@@ -44,12 +44,17 @@ 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}}-${{github.event.inputs.tag || github.ref_name}}
+        if: github.event.inputs.tag == 'test'
       - name: Upload artifact
         uses: actions/upload-artifact@v4
         with:
           name: agent-${{matrix.architecture}}.tar
           path: agent-${{matrix.architecture}}.tar
-          
+        
   build-arm64:
     runs-on: ubuntu-24.04-arm 
     permissions:
@@ -87,9 +92,38 @@ jobs:
           name: agent-${{matrix.architecture}}.tar
           path: agent-${{matrix.architecture}}.tar
 
+  build-darwin:
+    runs-on: ubuntu-24.04
+    permissions:
+      contents: write
+    strategy:
+      matrix:
+        architecture: [amd64, arm64]
+    steps:
+      - name: Checkout
+        uses: actions/checkout@v3
+      - name: Set up Docker Buildx
+        uses: docker/setup-buildx-action@v2
+      - name: Build Darwin binary with Docker
+        run: |
+          docker build -f Dockerfile.cross \
+            --build-arg GOOS=darwin \
+            --build-arg GOARCH=${{matrix.architecture}} \
+            -t darwin-${{matrix.architecture}} .
+          CID=$(docker create darwin-${{matrix.architecture}})
+          docker cp ${CID}:/home/agent ./output-${{matrix.architecture}}
+          docker rm ${CID}
+      - name: Tar Darwin binary and clean
+        run: tar -cf agent-darwin-${{matrix.architecture}}.tar -C output-${{matrix.architecture}} . && rm -rf output-${{matrix.architecture}}
+      - name: Upload darwin artifact
+        uses: actions/upload-artifact@v4
+        with:
+          name: agent-darwin-${{matrix.architecture}}.tar
+          path: agent-darwin-${{matrix.architecture}}.tar
+
   create-manifest:
     runs-on: ubuntu-24.04
-    needs: [build-amd64, build-arm64]
+    needs: [build-amd64, build-arm64, build-darwin]
     steps:
       - name: Login to DockerHub
         uses: docker/login-action@v2

.github/workflows/pr-build.yml

@@ -72,4 +72,40 @@ jobs:
         with:
           name: agent-${{matrix.architecture}}.tar
           path: agent-${{matrix.architecture}}.tar
+          
+  build-darwin:
+    runs-on: macos-13
+    permissions:
+      contents: write
+    strategy:
+      matrix:
+        architecture: [arm64]
+    steps:
+      - name: Setup Docker on macOS
+        uses: douglascamata/setup-docker-macos-action@v1.0.0
+      - name: Login to DockerHub
+        uses: docker/login-action@v2
+        with:
+          username: ${{ secrets.DOCKER_USERNAME }}
+          password: ${{ secrets.DOCKER_PASSWORD }}
+      - name: Checkout
+        uses: actions/checkout@v3
+      - uses: benjlevesque/short-sha@v2.1
+        id: short-sha
+        with:
+          length: 7
+      - name: Build Darwin binary with Docker
+        run: |
+          docker build -f Dockerfile.arm64 \
+            -t darwin-${{matrix.architecture}} .
+          CID=$(docker create darwin-${{matrix.architecture}})
+          docker cp ${CID}:/home/agent ./output-${{matrix.architecture}}
+          docker rm ${CID}
+      - name: Tar Darwin binary and clean
+        run: tar -cf agent-darwin-${{matrix.architecture}}.tar -C output-${{matrix.architecture}} . && rm -rf output-${{matrix.architecture}}
+      - name: Upload darwin artifact
+        uses: actions/upload-artifact@v4
+        with:
+          name: agent-darwin-${{matrix.architecture}}.tar
+          path: agent-darwin-${{matrix.architecture}}.tar
 

.github/workflows/pr-description.yaml

@@ -2,11 +2,6 @@ name: Autofill PR description
 
 on: pull_request
 
-env:
-  ORGANIZATION: uugai
-  PROJECT: ${{ github.event.repository.name }}
-  PR_NUMBER: ${{ github.event.number }}
-  
 jobs:
   openai-pr-description:
     runs-on: ubuntu-22.04
@@ -21,6 +16,4 @@ jobs:
           azure_openai_api_key: ${{ secrets.AZURE_OPENAI_API_KEY }}
           azure_openai_endpoint: ${{ secrets.AZURE_OPENAI_ENDPOINT }}
           azure_openai_version: ${{ secrets.AZURE_OPENAI_VERSION }}
-          openai_model: ${{ secrets.OPENAI_MODEL }}
-          pull_request_url: https://pr${{ env.PR_NUMBER }}.api.kerberos.lol
           overwrite_description: true

Dockerfile

@@ -32,6 +32,9 @@ RUN cat /go/src/github.com/kerberos-io/agent/machinery/version
 ##################
 # Build Machinery
 
+ARG GOOS=linux
+ENV GOOS=${GOOS}
+
 RUN cd /go/src/github.com/kerberos-io/agent/machinery && \
 	go mod download && \
 	go build -tags timetzdata,netgo,osusergo --ldflags '-s -w -extldflags "-static -latomic"' main.go && \

Dockerfile.arm64

@@ -32,6 +32,9 @@ RUN cat /go/src/github.com/kerberos-io/agent/machinery/version
 ##################
 # Build Machinery
 
+ARG GOOS=linux
+ENV GOOS=${GOOS}
+
 RUN cd /go/src/github.com/kerberos-io/agent/machinery && \
 	go mod download && \
 	go build -tags timetzdata,netgo,osusergo --ldflags '-s -w -extldflags "-static -latomic"' main.go && \

machinery/main.go

@@ -101,35 +101,31 @@ 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
-			}
-			onvif.Discover(timeout)
+		// 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)
+
 	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 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)
+		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
+		}
+
+		utils.Decrypt(flag.Arg(0), symmetricKey)
 
 	case "run":
 		{
@@ -217,8 +213,6 @@ 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/capture/gortsplib.go

@@ -85,8 +85,12 @@ type Golibrtsp struct {
 
 	Streams []packets.Stream
 
-	// Per-stream FPS calculation (keyed by stream index)
-	fpsTrackers map[int8]*fpsTracker
+	// FPS calculation fields
+	lastFrameTime    time.Time
+	frameTimeBuffer  []time.Duration
+	frameBufferSize  int
+	frameBufferIndex int
+	fpsMutex         sync.Mutex
 
 	// I-frame interval tracking fields
 	packetsSinceLastKeyframe int
@@ -97,78 +101,6 @@ type Golibrtsp struct {
 	keyframeMutex            sync.Mutex
 }
 
-// fpsTracker holds per-stream state for PTS-based FPS calculation.
-// Each video stream (H264 / H265) gets its own tracker so PTS
-// samples from different codecs never interleave.
-type fpsTracker struct {
-	mu              sync.Mutex
-	lastPTS         time.Duration
-	hasPTS          bool
-	frameTimeBuffer []time.Duration
-	bufferSize      int
-	bufferIndex     int
-	cachedFPS       float64 // latest computed FPS
-}
-
-func newFPSTracker(bufferSize int) *fpsTracker {
-	return &fpsTracker{
-		frameTimeBuffer: make([]time.Duration, bufferSize),
-		bufferSize:      bufferSize,
-	}
-}
-
-// update records a new PTS sample and returns the latest FPS estimate.
-// It must be called once per complete decoded frame (after Decode()
-// succeeds), not on every RTP packet fragment.
-func (ft *fpsTracker) update(pts time.Duration) float64 {
-	ft.mu.Lock()
-	defer ft.mu.Unlock()
-
-	if !ft.hasPTS {
-		ft.lastPTS = pts
-		ft.hasPTS = true
-		return 0
-	}
-
-	interval := pts - ft.lastPTS
-	ft.lastPTS = pts
-
-	// Skip invalid intervals (zero, negative, or very large which
-	// indicate a PTS discontinuity or wrap).
-	if interval <= 0 || interval > 5*time.Second {
-		return ft.cachedFPS
-	}
-
-	ft.frameTimeBuffer[ft.bufferIndex] = interval
-	ft.bufferIndex = (ft.bufferIndex + 1) % ft.bufferSize
-
-	var totalInterval time.Duration
-	validSamples := 0
-	for _, iv := range ft.frameTimeBuffer {
-		if iv > 0 {
-			totalInterval += iv
-			validSamples++
-		}
-	}
-	if validSamples == 0 {
-		return ft.cachedFPS
-	}
-	avgInterval := totalInterval / time.Duration(validSamples)
-	if avgInterval == 0 {
-		return ft.cachedFPS
-	}
-
-	ft.cachedFPS = float64(time.Second) / float64(avgInterval)
-	return ft.cachedFPS
-}
-
-// fps returns the most recent FPS estimate without recording a new sample.
-func (ft *fpsTracker) fps() float64 {
-	ft.mu.Lock()
-	defer ft.mu.Unlock()
-	return ft.cachedFPS
-}
-
 // Init function
 var H264FrameDecoder *Decoder
 var H265FrameDecoder *Decoder
@@ -616,17 +548,18 @@ func (g *Golibrtsp) Start(ctx context.Context, streamType string, queue *packets
 
 			if len(rtppkt.Payload) > 0 {
 
-				// decode timestamps — validate each call separately
-				pts, okPTS := g.Client.PacketPTS(g.VideoH264Media, rtppkt)
-				pts2, okPTS2 := g.Client.PacketPTS2(g.VideoH264Media, rtppkt)
-				if !okPTS2 {
-					log.Log.Debug("capture.golibrtsp.Start(): unable to get PTS2 from PacketPTS2")
+				// decode timestamp
+				pts, ok := g.Client.PacketPTS(g.VideoH264Media, rtppkt)
+				pts2, ok := g.Client.PacketPTS2(g.VideoH264Media, rtppkt)
+				if !ok {
+					log.Log.Debug("capture.golibrtsp.Start(): " + "unable to get PTS")
 					return
 				}
 
-				// Extract access units from RTP packets.
-				// We need a complete access unit to determine whether
-				// this is a keyframe.
+				// Extract access units from RTP packets
+				// We need to do this, because the decoder expects a full
+				// access unit. Once we have a full access unit, we can
+				// decode it, and know if it's a keyframe or not.
 				au, errDecode := g.VideoH264Decoder.Decode(rtppkt)
 				if errDecode != nil {
 					if errDecode != rtph264.ErrNonStartingPacketAndNoPrevious && errDecode != rtph264.ErrMorePacketsNeeded {
@@ -635,18 +568,6 @@ func (g *Golibrtsp) Start(ctx context.Context, streamType string, queue *packets
 					return
 				}
 
-				// Frame is complete — update per-stream FPS from PTS.
-				if okPTS {
-					ft := g.fpsTrackers[g.VideoH264Index]
-					if ft == nil {
-						ft = newFPSTracker(30)
-						g.fpsTrackers[g.VideoH264Index] = ft
-					}
-					if ptsFPS := ft.update(pts); ptsFPS > 0 && ptsFPS <= 120 {
-						g.Streams[g.VideoH264Index].FPS = ptsFPS
-					}
-				}
-
 				// We'll need to read out a few things.
 				// prepend an AUD. This is required by some players
 				filteredAU = [][]byte{
@@ -657,10 +578,8 @@ func (g *Golibrtsp) Start(ctx context.Context, streamType string, queue *packets
 				nonIDRPresent := false
 				idrPresent := false
 
-				var naluTypes []string
 				for _, nalu := range au {
 					typ := h264.NALUType(nalu[0] & 0x1F)
-					naluTypes = append(naluTypes, fmt.Sprintf("%s(%d,sz=%d)", typ.String(), int(typ), len(nalu)))
 					switch typ {
 					case h264.NALUTypeAccessUnitDelimiter:
 						continue
@@ -707,11 +626,6 @@ func (g *Golibrtsp) Start(ctx context.Context, streamType string, queue *packets
 					return
 				}
 
-				if idrPresent {
-					log.Log.Debug(fmt.Sprintf("capture.golibrtsp.Start(%s): IDR frame NALUs: [%s]",
-						streamType, fmt.Sprintf("%v", naluTypes)))
-				}
-
 				// Convert to packet.
 				enc, err := h264.AnnexBMarshal(filteredAU)
 				if err != nil {
@@ -737,11 +651,7 @@ func (g *Golibrtsp) Start(ctx context.Context, streamType string, queue *packets
 				keyframeInterval := g.trackKeyframeInterval(idrPresent)
 				if idrPresent && keyframeInterval > 0 {
 					avgInterval := g.getAverageKeyframeInterval()
-					fps := g.Streams[g.VideoH264Index].FPS
-					if fps <= 0 {
-						fps = 25.0 // Default fallback FPS
-					}
-					gopDuration := float64(keyframeInterval) / fps
+					gopDuration := float64(keyframeInterval) / g.Streams[g.VideoH265Index].FPS
 					gopSize := int(avgInterval) // Store GOP size in a separate variable
 					g.Streams[g.VideoH264Index].GopSize = gopSize
 					log.Log.Debug(fmt.Sprintf("capture.golibrtsp.Start(%s): Keyframe interval=%d packets, Avg=%.1f, GOP=%.1fs, GOPSize=%d",
@@ -806,17 +716,18 @@ func (g *Golibrtsp) Start(ctx context.Context, streamType string, queue *packets
 
 			if len(rtppkt.Payload) > 0 {
 
-				// decode timestamps — validate each call separately
-				pts, okPTS := g.Client.PacketPTS(g.VideoH265Media, rtppkt)
-				pts2, okPTS2 := g.Client.PacketPTS2(g.VideoH265Media, rtppkt)
-				if !okPTS2 {
-					log.Log.Debug("capture.golibrtsp.Start(): unable to get PTS")
+				// decode timestamp
+				pts, ok := g.Client.PacketPTS(g.VideoH265Media, rtppkt)
+				pts2, ok := g.Client.PacketPTS2(g.VideoH265Media, rtppkt)
+				if !ok {
+					log.Log.Debug("capture.golibrtsp.Start(): " + "unable to get PTS")
 					return
 				}
 
-				// Extract access units from RTP packets.
-				// We need a complete access unit to determine whether
-				// this is a keyframe.
+				// Extract access units from RTP packets
+				// We need to do this, because the decoder expects a full
+				// access unit. Once we have a full access unit, we can
+				// decode it, and know if it's a keyframe or not.
 				au, errDecode := g.VideoH265Decoder.Decode(rtppkt)
 				if errDecode != nil {
 					if errDecode != rtph265.ErrNonStartingPacketAndNoPrevious && errDecode != rtph265.ErrMorePacketsNeeded {
@@ -825,18 +736,6 @@ func (g *Golibrtsp) Start(ctx context.Context, streamType string, queue *packets
 					return
 				}
 
-				// Frame is complete — update per-stream FPS from PTS.
-				if okPTS {
-					ft := g.fpsTrackers[g.VideoH265Index]
-					if ft == nil {
-						ft = newFPSTracker(30)
-						g.fpsTrackers[g.VideoH265Index] = ft
-					}
-					if ptsFPS := ft.update(pts); ptsFPS > 0 && ptsFPS <= 120 {
-						g.Streams[g.VideoH265Index].FPS = ptsFPS
-					}
-				}
-
 				filteredAU = [][]byte{
 					{byte(h265.NALUType_AUD_NUT) << 1, 1, 0x50},
 				}
@@ -897,11 +796,7 @@ func (g *Golibrtsp) Start(ctx context.Context, streamType string, queue *packets
 				keyframeInterval := g.trackKeyframeInterval(isRandomAccess)
 				if isRandomAccess && keyframeInterval > 0 {
 					avgInterval := g.getAverageKeyframeInterval()
-					fps := g.Streams[g.VideoH265Index].FPS
-					if fps <= 0 {
-						fps = 25.0 // Default fallback FPS
-					}
-					gopDuration := float64(keyframeInterval) / fps
+					gopDuration := float64(keyframeInterval) / g.Streams[g.VideoH265Index].FPS
 					gopSize := int(avgInterval) // Store GOP size in a separate variable
 					g.Streams[g.VideoH265Index].GopSize = gopSize
 					log.Log.Debug(fmt.Sprintf("capture.golibrtsp.Start(%s): Keyframe interval=%d packets, Avg=%.1f, GOP=%.1fs, GOPSize=%d",
@@ -1284,11 +1179,10 @@ func WriteMPEG4Audio(forma *format.MPEG4Audio, aus [][]byte) ([]byte, error) {
 
 // Initialize FPS calculation buffers
 func (g *Golibrtsp) initFPSCalculation() {
-	// Ensure the per-stream FPS trackers map exists. Individual trackers
-	// can be created lazily when a given stream index is first used.
-	if g.fpsTrackers == nil {
-		g.fpsTrackers = make(map[int8]*fpsTracker)
-	}
+	g.frameBufferSize = 30 // Store last 30 frame intervals
+	g.frameTimeBuffer = make([]time.Duration, g.frameBufferSize)
+	g.frameBufferIndex = 0
+	g.lastFrameTime = time.Time{}
 
 	// Initialize I-frame interval tracking
 	g.keyframeBufferSize = 10 // Store last 10 keyframe intervals
@@ -1298,11 +1192,50 @@ func (g *Golibrtsp) initFPSCalculation() {
 	g.lastKeyframePacketCount = 0
 }
 
-// Get enhanced FPS information from SPS with fallback to PTS-based calculation.
-// The PTS-based FPS is computed per completed frame via fpsTracker.update(),
-// so by the time this is called we already have a good estimate.
+// Calculate FPS from frame timestamps
+func (g *Golibrtsp) calculateFPSFromTimestamps() float64 {
+	g.fpsMutex.Lock()
+	defer g.fpsMutex.Unlock()
+
+	if g.lastFrameTime.IsZero() {
+		g.lastFrameTime = time.Now()
+		return 0
+	}
+
+	now := time.Now()
+	interval := now.Sub(g.lastFrameTime)
+	g.lastFrameTime = now
+
+	// Store the interval
+	g.frameTimeBuffer[g.frameBufferIndex] = interval
+	g.frameBufferIndex = (g.frameBufferIndex + 1) % g.frameBufferSize
+
+	// Calculate average FPS from stored intervals
+	var totalInterval time.Duration
+	validSamples := 0
+
+	for _, interval := range g.frameTimeBuffer {
+		if interval > 0 {
+			totalInterval += interval
+			validSamples++
+		}
+	}
+
+	if validSamples == 0 {
+		return 0
+	}
+
+	avgInterval := totalInterval / time.Duration(validSamples)
+	if avgInterval == 0 {
+		return 0
+	}
+
+	return float64(time.Second) / float64(avgInterval)
+}
+
+// Get enhanced FPS information from SPS with fallback
 func (g *Golibrtsp) getEnhancedFPS(sps *h264.SPS, streamIndex int8) float64 {
-	// First try to get FPS from SPS VUI parameters
+	// First try to get FPS from SPS
 	spsFPS := sps.FPS()
 
 	// Check if SPS FPS is reasonable (between 1 and 120 fps)
@@ -1311,13 +1244,11 @@ func (g *Golibrtsp) getEnhancedFPS(sps *h264.SPS, streamIndex int8) float64 {
 		return spsFPS
 	}
 
-	// Fallback to PTS-based FPS (already calculated per-frame)
-	if ft := g.fpsTrackers[streamIndex]; ft != nil {
-		ptsFPS := ft.fps()
-		if ptsFPS > 0 && ptsFPS <= 120 {
-			log.Log.Debug(fmt.Sprintf("capture.golibrtsp.getEnhancedFPS(): PTS FPS: %.2f", ptsFPS))
-			return ptsFPS
-		}
+	// Fallback to timestamp-based calculation
+	timestampFPS := g.calculateFPSFromTimestamps()
+	if timestampFPS > 0 && timestampFPS <= 120 {
+		log.Log.Debug(fmt.Sprintf("capture.golibrtsp.getEnhancedFPS(): Timestamp FPS: %.2f", timestampFPS))
+		return timestampFPS
 	}
 
 	// Return SPS FPS even if it seems unreasonable, or default

machinery/src/capture/main.go

@@ -280,7 +280,7 @@ func HandleRecordStream(queue *packets.Queue, configDirectory string, configurat
 					vpsNALUS := configuration.Config.Capture.IPCamera.VPSNALUs
 
 					// Create a video file, and set the dimensions.
-					mp4Video = video.NewMP4(fullName, spsNALUS, ppsNALUS, vpsNALUS, configuration.Config.Capture.MaxLengthRecording)
+					mp4Video = video.NewMP4(fullName, spsNALUS, ppsNALUS, vpsNALUS)
 					mp4Video.SetWidth(width)
 					mp4Video.SetHeight(height)
 
@@ -500,7 +500,7 @@ func HandleRecordStream(queue *packets.Queue, configDirectory string, configurat
 				vpsNALUS := configuration.Config.Capture.IPCamera.VPSNALUs
 
 				// Create a video file, and set the dimensions.
-				mp4Video := video.NewMP4(fullName, spsNALUS, ppsNALUS, vpsNALUS, configuration.Config.Capture.MaxLengthRecording)
+				mp4Video := video.NewMP4(fullName, spsNALUS, ppsNALUS, vpsNALUS)
 				mp4Video.SetWidth(width)
 				mp4Video.SetHeight(height)
 

machinery/src/cloud/Cloud.go

@@ -804,12 +804,6 @@ 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

@@ -123,6 +123,7 @@ 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/video/mp4.go

@@ -22,77 +22,52 @@ import (
 
 var LastPTS uint64 = 0 // Last PTS for the current segment
 
-// FragmentDurationMs is the target duration for each fragment in milliseconds.
-// Fragments will be flushed at the first keyframe after this duration has elapsed,
-// resulting in ~3 second fragments (assuming a typical GOP interval).
-const FragmentDurationMs = 3000
-
 type MP4 struct {
 	// FileName is the name of the file
-	FileName                string
-	width                   int
-	height                  int
-	Segments                []*mp4ff.MediaSegment // List of media segments
-	Segment                 *mp4ff.MediaSegment
-	MultiTrackFragment      *mp4ff.Fragment
-	TrackIDs                []uint32
-	FileWriter              *os.File
-	Writer                  *bufio.Writer
-	SegmentCount            int
-	SampleCount             int
-	StartPTS                uint64
-	VideoTotalDuration      uint64
-	AudioTotalDuration      uint64
-	AudioPTS                uint64
-	Start                   bool
-	SPSNALUs                [][]byte // SPS NALUs for H264
-	PPSNALUs                [][]byte // PPS NALUs for H264
-	VPSNALUs                [][]byte // VPS NALUs for H264
-	FreeBoxSize             int64
-	FragmentStartRawPTS     uint64            // Raw PTS for timing when to flush fragments
-	FragmentStartDTS        uint64            // Accumulated VideoTotalDuration at fragment start (matches tfdt)
-	MoofBoxes               int64             // Number of moof boxes in the file
-	MoofBoxSizes            []int64           // Sizes of each moof box
-	SegmentDurations        []uint64          // Duration of each segment in timescale units
-	SegmentBaseDecTimes     []uint64          // Base decode time of each segment
-	StartTime               uint64            // Start time of the MP4 file
-	VideoTrackName          string            // Name of the video track
-	VideoTrack              int               // Track ID for the video track
-	AudioTrackName          string            // Name of the audio track
-	AudioTrack              int               // Track ID for the audio track
-	VideoFullSample         *mp4ff.FullSample // Full sample for video track
-	AudioFullSample         *mp4ff.FullSample // Full sample for audio track
-	LastAudioSampleDTS      uint64            // Last PTS for audio sample
-	LastVideoSampleDTS      uint64            // Last PTS for video sample
-	SampleType              string            // Type of the sample (e.g., "video", "audio", "subtitle")
-	TotalKeyframesReceived  int               // Total keyframes received by AddSampleToTrack
-	TotalKeyframesWritten   int               // Total keyframes written to trun boxes
-	FragmentKeyframeCount   int               // Keyframes in the current fragment
-	PendingSampleIsKeyframe bool              // Whether the pending video sample is a keyframe
+	FileName           string
+	width              int
+	height             int
+	Segments           []*mp4ff.MediaSegment // List of media segments
+	Segment            *mp4ff.MediaSegment
+	MultiTrackFragment *mp4ff.Fragment
+	TrackIDs           []uint32
+	FileWriter         *os.File
+	Writer             *bufio.Writer
+	SegmentCount       int
+	SampleCount        int
+	StartPTS           uint64
+	VideoTotalDuration uint64
+	AudioTotalDuration uint64
+	AudioPTS           uint64
+	Start              bool
+	SPSNALUs           [][]byte // SPS NALUs for H264
+	PPSNALUs           [][]byte // PPS NALUs for H264
+	VPSNALUs           [][]byte // VPS NALUs for H264
+	FreeBoxSize        int64
+	MoofBoxes          int64             // Number of moof boxes in the file
+	MoofBoxSizes       []int64           // Sizes of each moof box
+	StartTime          uint64            // Start time of the MP4 file
+	VideoTrackName     string            // Name of the video track
+	VideoTrack         int               // Track ID for the video track
+	AudioTrackName     string            // Name of the audio track
+	AudioTrack         int               // Track ID for the audio track
+	VideoFullSample    *mp4ff.FullSample // Full sample for video track
+	AudioFullSample    *mp4ff.FullSample // Full sample for audio track
+	LastAudioSampleDTS uint64            // Last PTS for audio sample
+	LastVideoSampleDTS uint64            // Last PTS for video sample
+	SampleType         string            // Type of the sample (e.g., "video", "audio", "subtitle")
 }
 
-// NewMP4 creates a new MP4 object.
-// maxDurationSec is the maximum expected recording duration in seconds,
-// used to calculate the free-box placeholder size for ftyp+moov+sidx.
-func NewMP4(fileName string, spsNALUs [][]byte, ppsNALUs [][]byte, vpsNALUs [][]byte, maxDurationSec int64) *MP4 {
+// NewMP4 creates a new MP4 object
+func NewMP4(fileName string, spsNALUs [][]byte, ppsNALUs [][]byte, vpsNALUs [][]byte) *MP4 {
 
-	// Calculate the placeholder size needed at the start of the file.
-	// Components:
-	//   ftyp:  ~32 bytes
-	//   moov:  ~1500 bytes (mvhd + mvex + video trak + audio trak + UUID)
-	//   sidx:  24 bytes fixed + 12 bytes per segment reference
-	// Segments are ~FragmentDurationMs each, so:
-	//   numSegments = ceil(maxDurationSec * 1000 / FragmentDurationMs) + 1 (safety margin)
-	//   sidxSize    = 24 + 12 * numSegments
-	baseSize := int64(2560) // ftyp + moov + extra headroom for large UUID signatures
-	numSegments := int64(0)
-	if maxDurationSec > 0 {
-		// Use integer ceiling division to avoid underestimating the number of segments.
-		numSegments = ((maxDurationSec*1000)+FragmentDurationMs-1)/FragmentDurationMs + 1
-	}
-	sidxSize := int64(24 + 12*numSegments)
-	freeBoxSize := int(baseSize + sidxSize)
+	init := mp4ff.NewMP4Init()
+
+	// Add a free box to the init segment
+	// Prepend a free box to the init segment with a size of 4096 bytes, so we can overwrite it later with the actual init segment.
+	freeBoxSize := 4096
 	free := mp4ff.NewFreeBox(make([]byte, freeBoxSize))
+	init.AddChild(free)
 
 	// Create a writer
 	ofd, err := os.Create(fileName)
@@ -102,15 +77,16 @@ func NewMP4(fileName string, spsNALUs [][]byte, ppsNALUs [][]byte, vpsNALUs [][]
 	// Create a buffered writer
 	bufferedWriter := bufio.NewWriterSize(ofd, 64*1024) // 64KB buffer
 
-	// Write the free box placeholder at the start of the file
-	err = free.Encode(bufferedWriter)
+	// We will write the empty init segment to the file
+	// so we can overwrite it later with the actual init segment.
+	err = init.Encode(bufferedWriter)
 	if err != nil {
 	}
 
 	return &MP4{
 		FileName:    fileName,
 		StartTime:   uint64(time.Now().Unix()),
-		FreeBoxSize: int64(freeBoxSize) + 8, // payload + 8 byte box header
+		FreeBoxSize: int64(freeBoxSize),
 		FileWriter:  ofd,
 		Writer:      bufferedWriter,
 		SPSNALUs:    spsNALUs,
@@ -154,126 +130,42 @@ func (mp4 *MP4) AddAudioTrack(codec string) uint32 {
 func (mp4 *MP4) AddMediaSegment(segNr int) {
 }
 
-// flushPendingVideoSample writes the pending video sample to the current fragment.
-// If nextPTS is provided (non-zero), it calculates duration from the PTS difference.
-// If nextPTS is 0 (e.g., at Close time), it uses the last known duration.
-// Returns true if a sample was flushed, false if there was no pending sample.
-func (mp4 *MP4) flushPendingVideoSample(nextPTS uint64) bool {
-	if mp4.VideoFullSample == nil || mp4.MultiTrackFragment == nil {
-		return false
-	}
-
-	var duration uint64
-	if nextPTS > 0 && nextPTS > mp4.VideoFullSample.DecodeTime {
-		duration = nextPTS - mp4.VideoFullSample.DecodeTime
-	} else {
-		// No valid nextPTS (Close case) or PTS went backwards (jitter/discontinuity)
-		if nextPTS > 0 {
-			log.Log.Warning(fmt.Sprintf("mp4.flushPendingVideoSample(): video PTS went backwards or zero duration (nextPTS=%d, prevDTS=%d), using last known duration", nextPTS, mp4.VideoFullSample.DecodeTime))
-		}
-		duration = mp4.LastVideoSampleDTS
-		if duration == 0 {
-			duration = 33 // Default ~30fps frame duration
-		}
-	}
-
-	mp4.LastVideoSampleDTS = duration
-	mp4.VideoTotalDuration += duration
-	mp4.VideoFullSample.DecodeTime = mp4.VideoTotalDuration - duration
-	mp4.VideoFullSample.Sample.Dur = uint32(duration)
-
-	isKF := mp4.PendingSampleIsKeyframe
-	err := mp4.MultiTrackFragment.AddFullSampleToTrack(*mp4.VideoFullSample, uint32(mp4.VideoTrack))
-	if err != nil {
-		log.Log.Error("mp4.flushPendingVideoSample(): error adding sample: " + err.Error())
-	}
-	if isKF {
-		mp4.TotalKeyframesWritten++
-		mp4.FragmentKeyframeCount++
-		log.Log.Debug(fmt.Sprintf("mp4.flushPendingVideoSample(): KEYFRAME WRITTEN to trun - totalWritten=%d, fragmentKF=%d, flags=0x%08x, dur=%d, DTS=%d",
-			mp4.TotalKeyframesWritten, mp4.FragmentKeyframeCount, mp4.VideoFullSample.Sample.Flags, duration, mp4.VideoFullSample.DecodeTime))
-	}
-
-	mp4.VideoFullSample = nil
-	mp4.PendingSampleIsKeyframe = false
-	return true
-}
-
 func (mp4 *MP4) AddSampleToTrack(trackID uint32, isKeyframe bool, data []byte, pts uint64) error {
 
-	if isKeyframe && trackID == uint32(mp4.VideoTrack) {
-		mp4.TotalKeyframesReceived++
-		elapsedDbg := uint64(0)
-		if mp4.Start {
-			elapsedDbg = pts - mp4.FragmentStartRawPTS
-		}
-		log.Log.Debug(fmt.Sprintf("mp4.AddSampleToTrack(): KEYFRAME #%d received - PTS=%d, size=%d, elapsed=%dms, started=%t, segment=%d, fragKF=%d",
-			mp4.TotalKeyframesReceived, pts, len(data), elapsedDbg, mp4.Start, mp4.SegmentCount, mp4.FragmentKeyframeCount))
-	}
-
 	if isKeyframe {
 
-		// Determine whether to start a new fragment.
-		// We only flush at a keyframe boundary once at least FragmentDurationMs
-		// of content has been accumulated, resulting in ~3 second fragments.
-		elapsed := uint64(0)
+		// Write the segment to the file
 		if mp4.Start {
-			elapsed = pts - mp4.FragmentStartRawPTS
-		}
-		shouldFlush := !mp4.Start || elapsed >= FragmentDurationMs
-
-		if shouldFlush {
-			// Write the previous segment to the file
-			if mp4.Start {
-				// IMPORTANT: Add any pending video sample to the current segment BEFORE flushing.
-				// This ensures the segment contains all frames up to (but not including) this keyframe,
-				// and the new segment will start cleanly with this keyframe.
-				if trackID == uint32(mp4.VideoTrack) {
-					mp4.flushPendingVideoSample(pts)
-				}
-
-				log.Log.Debug(fmt.Sprintf("mp4.AddSampleToTrack(): FLUSHING segment #%d - keyframes_in_fragment=%d, totalKF_received=%d, totalKF_written=%d",
-					mp4.SegmentCount, mp4.FragmentKeyframeCount, mp4.TotalKeyframesReceived, mp4.TotalKeyframesWritten))
-				mp4.MoofBoxes = mp4.MoofBoxes + 1
-				mp4.MoofBoxSizes = append(mp4.MoofBoxSizes, int64(mp4.Segment.Size()))
-				// Track the segment's duration and base decode time for sidx.
-				// Use accumulated VideoTotalDuration which matches the tfdt values
-				// in the trun boxes, NOT raw PTS from the camera.
-				segDuration := mp4.VideoTotalDuration - mp4.FragmentStartDTS
-				mp4.SegmentDurations = append(mp4.SegmentDurations, segDuration)
-				mp4.SegmentBaseDecTimes = append(mp4.SegmentBaseDecTimes, mp4.FragmentStartDTS)
-				err := mp4.Segment.Encode(mp4.Writer)
-				if err != nil {
-					log.Log.Error("mp4.AddSampleToTrack(): error encoding segment: " + err.Error())
-				}
-				mp4.Segments = append(mp4.Segments, mp4.Segment)
-			}
-
-			mp4.Start = true
-
-			// Increment the segment count
-			mp4.SegmentCount = mp4.SegmentCount + 1
-
-			// Create a new media segment
-			seg := mp4ff.NewMediaSegment()
-
-			// Create a video fragment
-			multiTrackFragment, err := mp4ff.CreateMultiTrackFragment(uint32(mp4.SegmentCount), mp4.TrackIDs)
+			mp4.MoofBoxes = mp4.MoofBoxes + 1
+			mp4.MoofBoxSizes = append(mp4.MoofBoxSizes, int64(mp4.Segment.Size()))
+			err := mp4.Segment.Encode(mp4.Writer)
 			if err != nil {
-				log.Log.Error("mp4.AddSampleToTrack(): error creating multi track fragment: " + err.Error())
+				log.Log.Error("mp4.AddSampleToTrack(): error encoding segment: " + err.Error())
 			}
-			mp4.MultiTrackFragment = multiTrackFragment
-			seg.AddFragment(multiTrackFragment)
-
-			// Set to MP4 struct
-			mp4.Segment = seg
-
-			// Set the start PTS for the next segment
-			mp4.StartPTS = pts
-			mp4.FragmentStartRawPTS = pts
-			mp4.FragmentStartDTS = mp4.VideoTotalDuration
-			mp4.FragmentKeyframeCount = 0 // Reset keyframe counter for new fragment
+			mp4.Segments = append(mp4.Segments, mp4.Segment)
 		}
+
+		mp4.Start = true
+
+		// Increment the segment count
+		mp4.SegmentCount = mp4.SegmentCount + 1
+
+		// Create a new media segment
+		seg := mp4ff.NewMediaSegment()
+
+		// Create a video fragment
+		multiTrackFragment, err := mp4ff.CreateMultiTrackFragment(uint32(mp4.SegmentCount), mp4.TrackIDs) // Assuming 1 for video track and 2 for audio track
+		if err != nil {
+			log.Log.Error("mp4.AddSampleToTrack(): error creating multi track fragment: " + err.Error())
+		}
+		mp4.MultiTrackFragment = multiTrackFragment
+		seg.AddFragment(multiTrackFragment)
+
+		// Set to MP4 struct
+		mp4.Segment = seg
+
+		// Set the start PTS for the next segment
+		mp4.StartPTS = pts
 	}
 
 	if mp4.Start {
@@ -290,10 +182,18 @@ func (mp4 *MP4) AddSampleToTrack(trackID uint32, isKeyframe bool, data []byte, p
 			}
 
 			if err == nil {
-				// Flush previous pending sample before storing the new one
 				if mp4.VideoFullSample != nil {
-					log.Log.Debug("Adding sample to track " + fmt.Sprintf("%d, PTS: %d, size: %d, Keyframe: %t", trackID, pts, len(lengthPrefixed), isKeyframe))
-					mp4.flushPendingVideoSample(pts)
+					duration := pts - mp4.VideoFullSample.DecodeTime
+					log.Log.Debug("Adding sample to track " + fmt.Sprintf("%d, PTS: %d, Duration: %d, size: %d, Keyframe: %t", trackID, pts, duration, len(lengthPrefixed), isKeyframe))
+
+					mp4.LastVideoSampleDTS = duration
+					mp4.VideoTotalDuration += duration
+					mp4.VideoFullSample.DecodeTime = mp4.VideoTotalDuration - duration
+					mp4.VideoFullSample.Sample.Dur = uint32(duration)
+					err := mp4.MultiTrackFragment.AddFullSampleToTrack(*mp4.VideoFullSample, trackID)
+					if err != nil {
+						log.Log.Error("mp4.AddSampleToTrack(): error adding sample to track " + fmt.Sprintf("%d: %v", trackID, err))
+					}
 				}
 
 				// Set the sample data
@@ -310,7 +210,6 @@ func (mp4 *MP4) AddSampleToTrack(trackID uint32, isKeyframe bool, data []byte, p
 					CompositionTimeOffset: 0, // No composition time offset for video
 				}
 				mp4.VideoFullSample = &fullSample
-				mp4.PendingSampleIsKeyframe = isKeyframe
 				mp4.SampleType = "video"
 			}
 		} else if trackID == uint32(mp4.AudioTrack) {
@@ -360,54 +259,12 @@ func (mp4 *MP4) AddSampleToTrack(trackID uint32, isKeyframe bool, data []byte, p
 
 func (mp4 *MP4) Close(config *models.Config) {
 
-	log.Log.Info(fmt.Sprintf("mp4.Close(): KEYFRAME SUMMARY - totalReceived=%d, totalWritten=%d, segments=%d, lastFragmentKF=%d",
-		mp4.TotalKeyframesReceived, mp4.TotalKeyframesWritten, mp4.SegmentCount, mp4.FragmentKeyframeCount))
-
 	if mp4.VideoTotalDuration == 0 && mp4.AudioTotalDuration == 0 {
 		log.Log.Error("mp4.Close(): no video or audio samples added, cannot create MP4 file")
 	}
 
-	// Add final pending samples before closing
-	if mp4.Segment != nil {
-		// Add final video sample if pending (pass 0 as nextPTS to use last known duration)
-		mp4.flushPendingVideoSample(0)
-
-		// Add final audio sample if pending
-		if mp4.AudioFullSample != nil && mp4.AudioTrack > 0 {
-			SplitAACFrame(mp4.AudioFullSample.Data, func(started bool, aac []byte) {
-				sampleToAdd := *mp4.AudioFullSample
-				dts := mp4.LastAudioSampleDTS
-				if dts == 0 {
-					dts = 1024 // Default AAC frame duration
-				}
-				mp4.AudioTotalDuration += dts
-				mp4.AudioPTS += dts
-				sampleToAdd.Data = aac[7:]
-				sampleToAdd.DecodeTime = mp4.AudioPTS - dts
-				sampleToAdd.Sample.Dur = uint32(dts)
-				sampleToAdd.Sample.Size = uint32(len(aac[7:]))
-				err := mp4.MultiTrackFragment.AddFullSampleToTrack(sampleToAdd, uint32(mp4.AudioTrack))
-				if err != nil {
-					log.Log.Error("mp4.Close(): error adding final audio sample: " + err.Error())
-				}
-			})
-			mp4.AudioFullSample = nil
-		}
-	}
-
 	// Encode the last segment
 	if mp4.Segment != nil {
-		// Track the last segment's size, duration and base decode time.
-		// Use accumulated VideoTotalDuration which matches tfdt values.
-		mp4.MoofBoxes = mp4.MoofBoxes + 1
-		mp4.MoofBoxSizes = append(mp4.MoofBoxSizes, int64(mp4.Segment.Size()))
-		lastSegDuration := mp4.VideoTotalDuration - mp4.FragmentStartDTS
-		if lastSegDuration == 0 {
-			lastSegDuration = mp4.LastVideoSampleDTS
-		}
-		mp4.SegmentDurations = append(mp4.SegmentDurations, lastSegDuration)
-		mp4.SegmentBaseDecTimes = append(mp4.SegmentBaseDecTimes, mp4.FragmentStartDTS)
-
 		err := mp4.Segment.Encode(mp4.Writer)
 		if err != nil {
 			log.Log.Error("mp4.Close(): error encoding last segment: " + err.Error())
@@ -415,14 +272,10 @@ func (mp4 *MP4) Close(config *models.Config) {
 	}
 
 	mp4.Writer.Flush()
-	// Ensure all segment data is on disk before we overwrite the placeholder at offset 0.
-	if err := mp4.FileWriter.Sync(); err != nil {
-		log.Log.Error("mp4.Close(): error syncing file: " + err.Error())
-	}
+	defer mp4.FileWriter.Close()
 
 	// Now we have all the moof and mdat boxes written to the file.
-	// We build the ftyp + moov init segment and write it at the start,
-	// overwriting the free box placeholder we reserved in NewMP4.
+	// We can now generate the ftyp and moov boxes, and replace it with the free box we added earlier (size of 2048 bytes).
 	init := mp4ff.NewMP4Init()
 
 	// Create a new ftyp box
@@ -463,10 +316,8 @@ func (mp4 *MP4) Close(config *models.Config) {
 		if err != nil {
 		}
 		init.Moov.Traks[0].Tkhd.Duration = mp4.VideoTotalDuration
-		init.Moov.Traks[0].Tkhd.Width = mp4ff.Fixed32(uint32(mp4.width) << 16)
-		init.Moov.Traks[0].Tkhd.Height = mp4ff.Fixed32(uint32(mp4.height) << 16)
 		init.Moov.Traks[0].Mdia.Hdlr.Name = "agent " + utils.VERSION
-		init.Moov.Traks[0].Mdia.Mdhd.Duration = mp4.VideoTotalDuration
+		//init.Moov.Traks[0].Mdia.Mdhd.Duration = mp4.VideoTotalDuration
 	case "H265", "HVC1":
 		init.AddEmptyTrack(videoTimescale, "video", "und")
 		includePS := true
@@ -474,10 +325,8 @@ func (mp4 *MP4) Close(config *models.Config) {
 		if err != nil {
 		}
 		init.Moov.Traks[0].Tkhd.Duration = mp4.VideoTotalDuration
-		init.Moov.Traks[0].Tkhd.Width = mp4ff.Fixed32(uint32(mp4.width) << 16)
-		init.Moov.Traks[0].Tkhd.Height = mp4ff.Fixed32(uint32(mp4.height) << 16)
 		init.Moov.Traks[0].Mdia.Hdlr.Name = "agent " + utils.VERSION
-		init.Moov.Traks[0].Mdia.Mdhd.Duration = mp4.VideoTotalDuration
+		//init.Moov.Traks[0].Mdia.Mdhd.Duration = mp4.VideoTotalDuration
 	}
 
 	// Try adding audio track if available
@@ -496,7 +345,7 @@ func (mp4 *MP4) Close(config *models.Config) {
 		}
 		init.Moov.Traks[1].Tkhd.Duration = mp4.AudioTotalDuration
 		init.Moov.Traks[1].Mdia.Hdlr.Name = "agent " + utils.VERSION
-		init.Moov.Traks[1].Mdia.Mdhd.Duration = mp4.AudioTotalDuration
+		//init.Moov.Traks[1].Mdia.Mdhd.Duration = mp4.AudioTotalDuration
 	}
 
 	// Try adding subtitle track if available
@@ -574,94 +423,126 @@ func (mp4 *MP4) Close(config *models.Config) {
 		}
 	}
 
-	// Build a Segment Index (sidx) box so players can seek directly to any
-	// fragment without scanning the entire file.
-	if len(mp4.SegmentDurations) > 0 {
-		sidx := &mp4ff.SidxBox{
-			Version:                  1,
-			Flags:                    0,
-			ReferenceID:              uint32(mp4.VideoTrack),
-			Timescale:                videoTimescale,
-			EarliestPresentationTime: 0,
-			FirstOffset:              0,
-			SidxRefs:                 make([]mp4ff.SidxRef, 0, len(mp4.SegmentDurations)),
-		}
-		for i, dur := range mp4.SegmentDurations {
-			sidx.SidxRefs = append(sidx.SidxRefs, mp4ff.SidxRef{
-				ReferenceType:      0, // media reference
-				ReferencedSize:     uint32(mp4.MoofBoxSizes[i]),
-				SubSegmentDuration: uint32(dur),
-				StartsWithSAP:      1,
-				SAPType:            1,
-			})
-		}
-		init.AddChild(sidx)
+	// We will also calculate the SIDX box, which is a segment index box that contains information about the segments in the file.
+	// This is useful for seeking in the file, and for streaming the file.
+	/*sidx := &mp4ff.SidxBox{
+		Version:                  0,
+		Flags:                    0,
+		ReferenceID:              0,
+		Timescale:                videoTimescale,
+		EarliestPresentationTime: 0,
+		FirstOffset:              0,
+		SidxRefs:                 make([]mp4ff.SidxRef, 0),
+	}
+	referenceTrak := init.Moov.Trak
+	trex, ok := init.Moov.Mvex.GetTrex(referenceTrak.Tkhd.TrackID)
+	if !ok {
+		// We have an issue.
 	}
 
-	// Encode the ftyp + moov + sidx into a buffer to measure the total size.
-	// Then compute the correct sidx.FirstOffset (the gap between the end of
-	// the sidx box and the first moof, occupied by the trailing free box)
-	// and re-encode with the corrected value.
-	var initBuf bytes.Buffer
-	if err := init.Encode(&initBuf); err != nil {
-		log.Log.Error("mp4.Close(): error encoding init segment: " + err.Error())
+	segDatas, err := findSegmentData(mp4.Segments, referenceTrak, trex)
+	if err != nil {
+		// We have an issue.
+	}
+	fillSidx(sidx, referenceTrak, segDatas, true)
+
+	// Add the SIDX box to the moov box
+	init.AddChild(sidx)*/
+
+	// Get a bit slice writer for the init segment
+	// Get a byte buffer of FreeBoxSize bytes to write the init segment
+	buffer := bytes.NewBuffer(make([]byte, 0))
+	init.Encode(buffer)
+
+	// The first FreeBoxSize bytes of the file is a free box, so we can read it and replace it with the moov box.
+	// The init box might not be FreeBoxSize bytes, so we need to read the first FreeBoxSize bytes and then replace it with the moov box.
+	// while the remaining bytes are for a new free box.
+	// Write the init segment at the beginning of the file, replacing the free box
+	if _, err := mp4.FileWriter.WriteAt(buffer.Bytes(), 0); err != nil {
 	}
 
-	initSize := int64(initBuf.Len())
-
-	// The sidx.FirstOffset is defined as the distance (in bytes) from the
-	// anchor point (first byte after the sidx box) to the first byte of
-	// the first referenced moof/mdat. Since sidx is the last box in init,
-	// the anchor point is at initSize, and the first moof is at FreeBoxSize.
-	if len(mp4.SegmentDurations) > 0 {
-		if mp4.FreeBoxSize < initSize {
-			// Avoid computing a negative offset and wrapping it to uint64.
-			log.Log.Error("mp4.Close(): FreeBoxSize is smaller than initSize; skipping sidx FirstOffset adjustment")
-		} else {
-			firstOffset := uint64(mp4.FreeBoxSize - initSize)
-			// Find the sidx we added and update its FirstOffset
-			for _, child := range init.Children {
-				if sidxBox, ok := child.(*mp4ff.SidxBox); ok {
-					sidxBox.FirstOffset = firstOffset
-					break
-				}
-			}
-			// Re-encode with the corrected FirstOffset (same size, no layout change)
-			initBuf.Reset()
-			if err := init.Encode(&initBuf); err != nil {
-				log.Log.Error("mp4.Close(): error re-encoding init segment: " + err.Error())
-			}
-			initSize = int64(initBuf.Len())
-		}
-	}
-
-	if initSize > mp4.FreeBoxSize {
-		log.Log.Error(fmt.Sprintf("mp4.Close(): init segment (%d bytes) exceeds reserved space (%d bytes), file may be corrupt", initSize, mp4.FreeBoxSize))
-	}
-
-	// Write the init segment at the beginning of the file, overwriting the free box placeholder.
-	if _, err := mp4.FileWriter.WriteAt(initBuf.Bytes(), 0); err != nil {
-		log.Log.Error("mp4.Close(): error writing init segment: " + err.Error())
-	}
-
-	// Fill any remaining reserved space with a new (smaller) free box so
-	// the byte offsets of the moof/mdat boxes that follow are preserved.
-	remainingSize := mp4.FreeBoxSize - initSize
-	if remainingSize >= 8 { // minimum box size is 8 bytes (header only)
-		newFree := mp4ff.NewFreeBox(make([]byte, remainingSize-8))
+	// Calculate the remaining size for the free box
+	remainingSize := mp4.FreeBoxSize - int64(buffer.Len())
+	if remainingSize > 0 {
+		newFreeBox := mp4ff.NewFreeBox(make([]byte, remainingSize))
 		var freeBuf bytes.Buffer
-		if err := newFree.Encode(&freeBuf); err != nil {
+		if err := newFreeBox.Encode(&freeBuf); err != nil {
 			log.Log.Error("mp4.Close(): error encoding free box: " + err.Error())
 		}
-		if _, err := mp4.FileWriter.WriteAt(freeBuf.Bytes(), initSize); err != nil {
+		if _, err := mp4.FileWriter.WriteAt(freeBuf.Bytes(), int64(buffer.Len())); err != nil {
 			log.Log.Error("mp4.Close(): error writing free box: " + err.Error())
 		}
 	}
+}
 
-	if err := mp4.FileWriter.Sync(); err != nil {
-		log.Log.Error("mp4.Close(): error syncing file: " + err.Error())
+type segData struct {
+	startPos         uint64
+	presentationTime uint64
+	baseDecodeTime   uint64
+	dur              uint32
+	size             uint32
+}
+
+func fillSidx(sidx *mp4ff.SidxBox, refTrak *mp4ff.TrakBox, segDatas []segData, nonZeroEPT bool) {
+	ept := uint64(0)
+	if nonZeroEPT {
+		ept = segDatas[0].presentationTime
 	}
-	mp4.FileWriter.Close()
+	sidx.Version = 1
+	sidx.Timescale = refTrak.Mdia.Mdhd.Timescale
+	sidx.ReferenceID = 1
+	sidx.EarliestPresentationTime = ept
+	sidx.FirstOffset = 0
+	sidx.SidxRefs = make([]mp4ff.SidxRef, 0, len(segDatas))
+
+	for _, segData := range segDatas {
+		size := segData.size
+		sidx.SidxRefs = append(sidx.SidxRefs, mp4ff.SidxRef{
+			ReferencedSize:     size,
+			SubSegmentDuration: segData.dur,
+			StartsWithSAP:      1,
+			SAPType:            1,
+		})
+	}
+}
+
+// findSegmentData returns a slice of segment media data using a reference track.
+func findSegmentData(segs []*mp4ff.MediaSegment, refTrak *mp4ff.TrakBox, trex *mp4ff.TrexBox) ([]segData, error) {
+	segDatas := make([]segData, 0, len(segs))
+	for _, seg := range segs {
+		var firstCompositionTimeOffest int64
+		dur := uint32(0)
+		var baseTime uint64
+		for fIdx, frag := range seg.Fragments {
+			for _, traf := range frag.Moof.Trafs {
+				tfhd := traf.Tfhd
+				if tfhd.TrackID == refTrak.Tkhd.TrackID { // Find track that gives sidx time values
+					if fIdx == 0 {
+						baseTime = traf.Tfdt.BaseMediaDecodeTime()
+					}
+					for i, trun := range traf.Truns {
+						trun.AddSampleDefaultValues(tfhd, trex)
+						samples := trun.GetSamples()
+						for j, sample := range samples {
+							if fIdx == 0 && i == 0 && j == 0 {
+								firstCompositionTimeOffest = int64(sample.CompositionTimeOffset)
+							}
+							dur += sample.Dur
+						}
+					}
+				}
+			}
+		}
+		sd := segData{
+			startPos:         seg.StartPos,
+			presentationTime: uint64(int64(baseTime) + firstCompositionTimeOffest),
+			baseDecodeTime:   baseTime,
+			dur:              dur,
+			size:             uint32(seg.Size()),
+		}
+		segDatas = append(segDatas, sd)
+	}
+	return segDatas, nil
 }
 
 // annexBToLengthPrefixed converts Annex B formatted H264 data (with start codes)

machinery/src/webrtc/main.go

@@ -1,7 +1,6 @@
 package webrtc
 
 import (
-	"context"
 	"encoding/base64"
 	"encoding/json"
 	"io"
@@ -23,104 +22,12 @@ import (
 	pionMedia "github.com/pion/webrtc/v4/pkg/media"
 )
 
-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
+var (
+	CandidatesMutex     sync.Mutex
+	CandidateArrays     map[string](chan string)
 	peerConnectionCount int64
-}
-
-// peerConnectionWrapper wraps a peer connection with additional metadata
-type peerConnectionWrapper struct {
-	conn      *pionWebRTC.PeerConnection
-	cancelCtx context.CancelFunc
-	done      chan struct{}
-	closeOnce sync.Once
-}
-
-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)
-}
+	peerConnections     map[string]*pionWebRTC.PeerConnection
+)
 
 type WebRTC struct {
 	Name                  string
@@ -139,7 +46,7 @@ func CreateWebRTC(name string, stunServers []string, turnServers []string, turnS
 		TurnServers:           turnServers,
 		TurnServersUsername:   turnServersUsername,
 		TurnServersCredential: turnServersCredential,
-		Timer:                 time.NewTimer(defaultTimeout),
+		Timer:                 time.NewTimer(time.Second * 10),
 	}
 }
 
@@ -161,14 +68,19 @@ func (w WebRTC) CreateOffer(sd []byte) pionWebRTC.SessionDescription {
 }
 
 func RegisterCandidates(key string, candidate models.ReceiveHDCandidatesPayload) {
-	ch := globalConnectionManager.GetOrCreateCandidateChannel(key)
-
-	log.Log.Info("webrtc.main.RegisterCandidates(): " + candidate.Candidate)
-	select {
-	case ch <- candidate.Candidate:
-	default:
-		log.Log.Info("webrtc.main.RegisterCandidates(): channel is full, dropping candidate")
+	// Set lock
+	CandidatesMutex.Lock()
+	_, ok := CandidateArrays[key]
+	if !ok {
+		CandidateArrays[key] = make(chan string, 100)
 	}
+	log.Log.Info("webrtc.main.HandleReceiveHDCandidates(): " + candidate.Candidate)
+	select {
+	case CandidateArrays[key] <- candidate.Candidate:
+	default:
+		log.Log.Info("webrtc.main.HandleReceiveHDCandidates(): channel is full.")
+	}
+	CandidatesMutex.Unlock()
 }
 
 func RegisterDefaultInterceptors(mediaEngine *pionWebRTC.MediaEngine, interceptorRegistry *interceptor.Registry) error {
@@ -195,7 +107,12 @@ func InitializeWebRTCConnection(configuration *models.Configuration, communicati
 
 	// We create a channel which will hold the candidates for this session.
 	sessionKey := config.Key + "/" + handshake.SessionID
-	candidateChannel := globalConnectionManager.GetOrCreateCandidateChannel(sessionKey)
+	CandidatesMutex.Lock()
+	_, ok := CandidateArrays[sessionKey]
+	if !ok {
+		CandidateArrays[sessionKey] = make(chan string, 100)
+	}
+	CandidatesMutex.Unlock()
 
 	// Set variables
 	hubKey := handshake.HubKey
@@ -261,128 +178,81 @@ func InitializeWebRTCConnection(configuration *models.Configuration, communicati
 
 		if err == nil && peerConnection != nil {
 
-			// 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")
+			if videoSender, err = peerConnection.AddTrack(videoTrack); err != nil {
+				log.Log.Error("webrtc.main.InitializeWebRTCConnection(): something went wrong while adding video track: " + err.Error())
 			}
-
 			// Read incoming RTCP packets
 			// Before these packets are returned they are processed by interceptors. For things
 			// like NACK this needs to be called.
-			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
-							}
-						}
+			go func() {
+				rtcpBuf := make([]byte, 1500)
+				for {
+					if _, _, rtcpErr := videoSender.Read(rtcpBuf); rtcpErr != nil {
+						return
 					}
-				}()
-			}
+				}
+			}()
 
 			var audioSender *pionWebRTC.RTPSender = nil
-			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
+			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
 			// Before these packets are returned they are processed by interceptors. For things
 			// like NACK this needs to be called.
-			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
-							}
-						}
+			go func() {
+				rtcpBuf := make([]byte, 1500)
+				for {
+					if _, _, rtcpErr := audioSender.Read(rtcpBuf); rtcpErr != nil {
+						return
 					}
-				}()
-			}
+				}
+			}()
 
 			peerConnection.OnConnectionStateChange(func(connectionState pionWebRTC.PeerConnectionState) {
-				log.Log.Info("webrtc.main.InitializeWebRTCConnection(): connection state changed to: " + connectionState.String())
-
-				switch connectionState {
-				case pionWebRTC.PeerConnectionStateDisconnected, pionWebRTC.PeerConnectionStateClosed:
-					wrapper.closeOnce.Do(func() {
-						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")
+				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) + ")")
 			})
 
 			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
-				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())
-						}
+				// 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())
 					}
+					CandidatesMutex.Unlock()
 				}
 			}()
 
@@ -400,56 +270,22 @@ 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{})
-				candateBinary, err := json.Marshal(candidateJSON)
+				candateJSON := candidate.ToJSON()
+				candateBinary, err := json.Marshal(candateJSON)
 				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.Error("webrtc.main.InitializeWebRTCConnection(): failed to marshal candidate: " + err.Error())
+					log.Log.Info("webrtc.main.InitializeWebRTCConnection(): something went wrong while marshalling candidate: " + err.Error())
 				}
 
 				// We'll send the candidate to the hub
@@ -469,8 +305,8 @@ func InitializeWebRTCConnection(configuration *models.Configuration, communicati
 				}
 			})
 
-			// Store peer connection in manager
-			globalConnectionManager.AddPeerConnection(handshake.SessionID, wrapper)
+			// Create a channel which will be used to send candidates to the other peer
+			peerConnections[handshake.SessionID] = peerConnection
 
 			if err == nil {
 				//  Create a config map
@@ -503,11 +339,7 @@ func InitializeWebRTCConnection(configuration *models.Configuration, communicati
 
 func NewVideoTrack(streams []packets.Stream) *pionWebRTC.TrackLocalStaticSample {
 	mimeType := pionWebRTC.MimeTypeH264
-	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
-	}
+	outboundVideoTrack, _ := pionWebRTC.NewTrackLocalStaticSample(pionWebRTC.RTPCodecCapability{MimeType: mimeType}, "video", "pion124")
 	return outboundVideoTrack
 }
 
@@ -522,245 +354,161 @@ func NewAudioTrack(streams []packets.Stream) *pionWebRTC.TrackLocalStaticSample
 			mimeType = pionWebRTC.MimeTypePCMA
 		}
 	}
-	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
-	}
+	outboundAudioTrack, _ := pionWebRTC.NewTrackLocalStaticSample(pionWebRTC.RTPCodecCapability{MimeType: mimeType}, "audio", "pion124")
 	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
 
-	// 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
+	// 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
+		}
 	}
 
-	// Detect available codecs
-	codecs := detectCodecs(rtspClient)
-
-	if !codecs.hasValidCodecs() {
-		log.Log.Error("webrtc.main.WriteToTrack(): no valid video or audio codec found")
-		return
-	}
-
-	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
+	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.")
 		}
 
-		// Update state from communication channels
-		updateStreamState(communication, state)
+		var cursorError error
+		var pkt packets.Packet
+		var lastAudioSample *pionMedia.Sample = nil
+		var lastVideoSample *pionMedia.Sample = nil
 
-		// Check if we should continue streaming
-		if !shouldContinueStreaming(config, state) {
-			state.start = false
-			state.receivedKeyFrame = false
-			continue
-		}
+		start := false
+		receivedKeyFrame := false
+		lastKeepAlive := "0"
+		peerCount := "0"
 
-		// Skip empty packets
-		if len(pkt.Data) == 0 || pkt.Data == nil {
-			state.receivedKeyFrame = false
-			continue
-		}
+		for cursorError == nil {
 
-		// Wait for first keyframe before processing
-		if !state.receivedKeyFrame {
-			if pkt.IsKeyFrame {
-				state.receivedKeyFrame = true
-			} else {
+			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
 			}
-		}
 
-		// Process video or audio packets
-		if pkt.IsVideo {
-			processVideoPacket(pkt, state, videoTrack, config)
-		} else if pkt.IsAudio {
-			processAudioPacket(pkt, state, audioTrack, codecs.hasAAC)
+			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
+			}
 		}
 	}
+
+	peerConnectionCount = 0
+	log.Log.Info("webrtc.main.WriteToTrack(): stop writing to track.")
 }