diff --git a/internal/desktop/pipeline.go b/internal/desktop/pipeline.go
index 97d24e4..a50448b 100644
--- a/internal/desktop/pipeline.go
+++ b/internal/desktop/pipeline.go
@@ -178,9 +178,10 @@ func (pm *PipelineManager) buildGStreamer(cfg PipelineConfig) (string, []string,
 	pipelineParts = append(pipelineParts, fmt.Sprintf("ximagesrc display-name=%s", cfg.Display))
 	pipelineParts = append(pipelineParts, "video/x-raw,framerate=30/1")
 	pipelineParts = append(pipelineParts, "videoconvert")
+	pipelineParts = append(pipelineParts, "video/x-raw,format=I420")
 
 	// 2. Encoder
-	encoderStr := "x264enc speed-preset=ultrafast tune=zerolatency bitrate=" + fmt.Sprintf("%d", cfg.Bitrate)
+	encoderStr := "x264enc speed-preset=ultrafast tune=zerolatency bitrate=" + fmt.Sprintf("%d", cfg.Bitrate) + " byte-stream=true key-int-max=30"
 	if cfg.UseGPU {
 		// Detect if nvcodec is present by calling gst-inspect-1.0 or simply attempting it.
 		// We'll default to nvh264enc.
@@ -189,6 +190,7 @@ func (pm *PipelineManager) buildGStreamer(cfg PipelineConfig) (string, []string,
 	pipelineParts = append(pipelineParts, encoderStr)
 
 	// 3. Payload and Sink
+	pipelineParts = append(pipelineParts, "video/x-h264,profile=baseline")
 	pipelineParts = append(pipelineParts, "rtph264pay config-interval=1 pt=96")
 	pipelineParts = append(pipelineParts, fmt.Sprintf("udpsink host=127.0.0.1 port=%d sync=false async=false", cfg.Port))
 
@@ -224,12 +226,16 @@ func (pm *PipelineManager) buildFFmpeg(cfg PipelineConfig) (string, []string, er
 			"-preset", "llhp", // low latency high quality
 			"-tune", "zerolatency",
 			"-g", "30",
+			"-pix_fmt", "yuv420p",
 		)
 	} else {
 		args = append(args,
 			"-c:v", "libx264",
 			"-preset", "ultrafast",
 			"-tune", "zerolatency",
+			"-profile:v", "baseline",
+			"-pix_fmt", "yuv420p",
+			"-x264-params", "repeat-headers=1",
 			"-g", "30",
 		)
 	}
diff --git a/internal/desktop/pipeline_test.go b/internal/desktop/pipeline_test.go
new file mode 100644
index 0000000..410d633
--- /dev/null
+++ b/internal/desktop/pipeline_test.go
@@ -0,0 +1,71 @@
+package desktop
+
+import (
+	"strings"
+	"testing"
+)
+
+func TestBuildGStreamerUsesBrowserFriendlyH264(t *testing.T) {
+	pm := NewPipelineManager()
+
+	tool, args, err := pm.buildGStreamer(PipelineConfig{
+		Display: ":11.0",
+		Port:    5004,
+		Bitrate: 2000,
+	})
+	if err != nil {
+		t.Fatalf("buildGStreamer returned error: %v", err)
+	}
+
+	if tool != "gst-launch-1.0" {
+		t.Fatalf("expected gst-launch-1.0, got %q", tool)
+	}
+	joined := strings.Join(args, " ")
+	for _, expected := range []string{
+		"video/x-raw,format=I420",
+		"x264enc",
+		"tune=zerolatency",
+		"key-int-max=30",
+		"byte-stream=true",
+		"video/x-h264,profile=baseline",
+		"rtph264pay",
+		"config-interval=1",
+	} {
+		if !strings.Contains(joined, expected) {
+			t.Fatalf("expected GStreamer args to contain %q, got %s", expected, joined)
+		}
+	}
+}
+
+func TestBuildFFmpegUsesBrowserFriendlyH264(t *testing.T) {
+	pm := NewPipelineManager()
+
+	tool, args, err := pm.buildFFmpeg(PipelineConfig{
+		Display: ":11.0",
+		Port:    5004,
+		Bitrate: 2000,
+		FPS:     30,
+		Width:   1280,
+		Height:  720,
+	})
+	if err != nil {
+		t.Fatalf("buildFFmpeg returned error: %v", err)
+	}
+
+	if tool != "ffmpeg" {
+		t.Fatalf("expected ffmpeg, got %q", tool)
+	}
+	joined := strings.Join(args, " ")
+	for _, expected := range []string{
+		"-c:v libx264",
+		"-tune zerolatency",
+		"-profile:v baseline",
+		"-pix_fmt yuv420p",
+		"-x264-params repeat-headers=1",
+		"-g 30",
+	} {
+		if !strings.Contains(joined, expected) {
+			t.Fatalf("expected FFmpeg args to contain %q, got %s", expected, joined)
+		}
+	}
+}
diff --git a/internal/desktop/webrtc.go b/internal/desktop/webrtc.go
index d3b3ac3..9ad8909 100644
--- a/internal/desktop/webrtc.go
+++ b/internal/desktop/webrtc.go
@@ -5,7 +5,10 @@ import (
 	"fmt"
 	"log"
 	"net"
+	"os"
 	"sync"
+	"sync/atomic"
+	"time"
 
 	"github.com/pion/webrtc/v4"
 )
@@ -17,6 +20,19 @@ type WebRTCServer struct {
 	inputInjector  *XdotoolInjector
 	mu             sync.Mutex
 	isClosed       bool
+	rtpPackets     uint64
+	rtpBytes       uint64
+	rtpWriteErrors uint64
+}
+
+func desktopWebRTCDiagnosticsEnabled() bool {
+	value := os.Getenv("XWORKMATE_DESKTOP_WEBRTC_DEBUG")
+	switch value {
+	case "0", "false", "FALSE", "off", "OFF", "no", "NO":
+		return false
+	default:
+		return true
+	}
 }
 
 func NewWebRTCServer(injector *XdotoolInjector) (*WebRTCServer, error) {
@@ -107,6 +123,36 @@ func (w *WebRTCServer) StartRTPReceiver(port int) error {
 	go func() {
 		buf := make([]byte, 2048)
 		log.Printf("WebRTC RTP receiver listening on UDP %s", addr)
+		statsDone := make(chan struct{})
+		if desktopWebRTCDiagnosticsEnabled() {
+			statsTicker := time.NewTicker(5 * time.Second)
+			defer statsTicker.Stop()
+			defer close(statsDone)
+			go func() {
+				var lastPackets uint64
+				var lastBytes uint64
+				for {
+					select {
+					case <-statsTicker.C:
+						packets := atomic.LoadUint64(&w.rtpPackets)
+						bytes := atomic.LoadUint64(&w.rtpBytes)
+						errors := atomic.LoadUint64(&w.rtpWriteErrors)
+						log.Printf(
+							"WebRTC RTP stats: packets=%d bytes=%d packetDelta=%d byteDelta=%d writeErrors=%d",
+							packets,
+							bytes,
+							packets-lastPackets,
+							bytes-lastBytes,
+							errors,
+						)
+						lastPackets = packets
+						lastBytes = bytes
+					case <-statsDone:
+						return
+					}
+				}
+			}()
+		}
 		for {
 			n, _, err := conn.ReadFrom(buf)
 			if err != nil {
@@ -118,6 +164,8 @@ func (w *WebRTCServer) StartRTPReceiver(port int) error {
 				}
 				break
 			}
+			atomic.AddUint64(&w.rtpPackets, 1)
+			atomic.AddUint64(&w.rtpBytes, uint64(n))
 
 			// Forward packet directly to WebRTC track (zero-copy)
 			w.mu.Lock()
@@ -126,6 +174,7 @@ func (w *WebRTCServer) StartRTPReceiver(port int) error {
 
 			if track != nil {
 				if _, err := track.Write(buf[:n]); err != nil {
+					atomic.AddUint64(&w.rtpWriteErrors, 1)
 					log.Printf("Failed to write RTP packet to track: %v", err)
 				}
 			}
@@ -214,4 +263,10 @@ func (w *WebRTCServer) Close() {
 	if pc != nil {
 		_ = pc.Close()
 	}
+	log.Printf(
+		"WebRTC RTP final stats: packets=%d bytes=%d writeErrors=%d",
+		atomic.LoadUint64(&w.rtpPackets),
+		atomic.LoadUint64(&w.rtpBytes),
+		atomic.LoadUint64(&w.rtpWriteErrors),
+	)
 }