WebRTC session exchange working

internal/monitoring/monitoring.go

@@ -15,15 +15,21 @@ type Options struct {
 }
 
 var (
-	// ViewerServed is the total amount of viewer page served
+	// WebViewerServed is the total amount of viewer page served
-	ViewerServed = promauto.NewCounter(prometheus.CounterOpts{
+	WebViewerServed = promauto.NewCounter(prometheus.CounterOpts{
-		Name: "ghostream_viewer_served_total",
+		Name: "ghostream_web_viewer_served_total",
 		Help: "The total amount of viewer served",
 	})
+
+	// WebSessions is the total amount of WebRTC session exchange
+	WebSessions = promauto.NewCounter(prometheus.CounterOpts{
+		Name: "ghostream_web_sessions_total",
+		Help: "The total amount of WebRTC sessions exchanged",
+	})
 )
 
-// ServeHTTP server that expose prometheus metrics
+// Serve monitoring server that expose prometheus metrics
-func ServeHTTP(cfg *Options) {
+func Serve(cfg *Options) {
 	mux := http.NewServeMux()
 	mux.Handle("/metrics", promhttp.Handler())
 	log.Printf("Monitoring HTTP server listening on %s", cfg.ListenAddress)

main.go

@@ -4,9 +4,11 @@ import (
 	"log"
 	"strings"
 
+	"github.com/pion/webrtc/v3"
 	"github.com/spf13/viper"
 	"gitlab.crans.org/nounous/ghostream/auth"
 	"gitlab.crans.org/nounous/ghostream/internal/monitoring"
+	"gitlab.crans.org/nounous/ghostream/stream"
 	"gitlab.crans.org/nounous/ghostream/web"
 )
 
@@ -68,15 +70,14 @@ func main() {
 	}
 	defer authBackend.Close()
 
-	// Start web server routine
+	// WebRTC session description channels
-	go func() {
+	remoteSdpChan := make(chan webrtc.SessionDescription)
-		web.ServeHTTP(&cfg.Web)
+	localSdpChan := make(chan webrtc.SessionDescription)
-	}()
 
-	// Start monitoring server routine
+	// Start stream, web and monitoring server
-	go func() {
+	go stream.Serve(remoteSdpChan, localSdpChan)
-		monitoring.ServeHTTP(&cfg.Monitoring)
+	go web.Serve(remoteSdpChan, localSdpChan, &cfg.Web)
-	}()
+	go monitoring.Serve(&cfg.Monitoring)
 
 	// Wait for routines
 	select {}

stream/stream.go

@@ -0,0 +1,207 @@
+package stream
+
+import (
+	"context"
+	"fmt"
+	"io"
+	"math/rand"
+	"os"
+	"time"
+
+	"github.com/pion/webrtc/v3"
+	"github.com/pion/webrtc/v3/pkg/media"
+	"github.com/pion/webrtc/v3/pkg/media/ivfreader"
+	"github.com/pion/webrtc/v3/pkg/media/oggreader"
+)
+
+const (
+	audioFileName = "output.ogg"
+	videoFileName = "output.ivf"
+)
+
+// Serve WebRTC media streaming server
+func Serve(remoteSdpChan chan webrtc.SessionDescription, localSdpChan chan webrtc.SessionDescription) {
+	// Assert that we have an audio or video file
+	_, err := os.Stat(videoFileName)
+	haveVideoFile := !os.IsNotExist(err)
+	_, err = os.Stat(audioFileName)
+	haveAudioFile := !os.IsNotExist(err)
+	if !haveAudioFile && !haveVideoFile {
+		panic("Could not find `" + audioFileName + "` or `" + videoFileName + "`")
+	}
+
+	// Passing client offer
+	offer := <-remoteSdpChan
+
+	// We make our own mediaEngine so we can place the sender's codecs in it.  This because we must use the
+	// dynamic media type from the sender in our answer. This is not required if we are the offerer
+	mediaEngine := webrtc.MediaEngine{}
+	if err = mediaEngine.PopulateFromSDP(offer); err != nil {
+		panic(err)
+	}
+
+	// Create a new RTCPeerConnection
+	api := webrtc.NewAPI(webrtc.WithMediaEngine(mediaEngine))
+	peerConnection, err := api.NewPeerConnection(webrtc.Configuration{
+		ICEServers: []webrtc.ICEServer{
+			{
+				URLs: []string{"stun:stun.l.google.com:19302"},
+			},
+		},
+	})
+	if err != nil {
+		panic(err)
+	}
+	iceConnectedCtx, iceConnectedCtxCancel := context.WithCancel(context.Background())
+
+	if haveVideoFile {
+		// Create a video track
+		videoTrack, addTrackErr := peerConnection.NewTrack(getPayloadType(mediaEngine, webrtc.RTPCodecTypeVideo, "VP8"), rand.Uint32(), "video", "pion")
+		if addTrackErr != nil {
+			panic(addTrackErr)
+		}
+		if _, addTrackErr = peerConnection.AddTrack(videoTrack); addTrackErr != nil {
+			panic(addTrackErr)
+		}
+
+		go func() {
+			// Open a IVF file and start reading using our IVFReader
+			file, ivfErr := os.Open(videoFileName)
+			if ivfErr != nil {
+				panic(ivfErr)
+			}
+
+			ivf, header, ivfErr := ivfreader.NewWith(file)
+			if ivfErr != nil {
+				panic(ivfErr)
+			}
+
+			// Wait for connection established
+			<-iceConnectedCtx.Done()
+
+			// Send our video file frame at a time. Pace our sending so we send it at the same speed it should be played back as.
+			// This isn't required since the video is timestamped, but we will such much higher loss if we send all at once.
+			sleepTime := time.Millisecond * time.Duration((float32(header.TimebaseNumerator)/float32(header.TimebaseDenominator))*1000)
+			for {
+				frame, _, ivfErr := ivf.ParseNextFrame()
+				if ivfErr == io.EOF {
+					fmt.Printf("All video frames parsed and sent")
+					os.Exit(0)
+				}
+
+				if ivfErr != nil {
+					panic(ivfErr)
+				}
+
+				time.Sleep(sleepTime)
+				if ivfErr = videoTrack.WriteSample(media.Sample{Data: frame, Samples: 90000}); ivfErr != nil {
+					panic(ivfErr)
+				}
+			}
+		}()
+	}
+
+	if haveAudioFile {
+		// Create a audio track
+		audioTrack, addTrackErr := peerConnection.NewTrack(getPayloadType(mediaEngine, webrtc.RTPCodecTypeAudio, "opus"), rand.Uint32(), "audio", "pion")
+		if addTrackErr != nil {
+			panic(addTrackErr)
+		}
+		if _, addTrackErr = peerConnection.AddTrack(audioTrack); addTrackErr != nil {
+			panic(addTrackErr)
+		}
+
+		go func() {
+			// Open a IVF file and start reading using our IVFReader
+			file, oggErr := os.Open(audioFileName)
+			if oggErr != nil {
+				panic(oggErr)
+			}
+
+			// Open on oggfile in non-checksum mode.
+			ogg, _, oggErr := oggreader.NewWith(file)
+			if oggErr != nil {
+				panic(oggErr)
+			}
+
+			// Wait for connection established
+			<-iceConnectedCtx.Done()
+
+			// Keep track of last granule, the difference is the amount of samples in the buffer
+			var lastGranule uint64
+			for {
+				pageData, pageHeader, oggErr := ogg.ParseNextPage()
+				if oggErr == io.EOF {
+					fmt.Printf("All audio pages parsed and sent")
+					os.Exit(0)
+				}
+
+				if oggErr != nil {
+					panic(oggErr)
+				}
+
+				// The amount of samples is the difference between the last and current timestamp
+				sampleCount := float64(pageHeader.GranulePosition - lastGranule)
+				lastGranule = pageHeader.GranulePosition
+
+				if oggErr = audioTrack.WriteSample(media.Sample{Data: pageData, Samples: uint32(sampleCount)}); oggErr != nil {
+					panic(oggErr)
+				}
+
+				// Convert seconds to Milliseconds, Sleep doesn't accept floats
+				time.Sleep(time.Duration((sampleCount/48000)*1000) * time.Millisecond)
+			}
+		}()
+	}
+
+	// Set the handler for ICE connection state
+	// This will notify you when the peer has connected/disconnected
+	peerConnection.OnICEConnectionStateChange(func(connectionState webrtc.ICEConnectionState) {
+		fmt.Printf("Connection State has changed %s \n", connectionState.String())
+		if connectionState == webrtc.ICEConnectionStateConnected {
+			iceConnectedCtxCancel()
+		}
+	})
+
+	// Set the remote SessionDescription
+	if err = peerConnection.SetRemoteDescription(offer); err != nil {
+		panic(err)
+	}
+
+	// Create answer
+	answer, err := peerConnection.CreateAnswer(nil)
+	if err != nil {
+		panic(err)
+	}
+
+	// Create channel that is blocked until ICE Gathering is complete
+	gatherComplete := webrtc.GatheringCompletePromise(peerConnection)
+
+	// Sets the LocalDescription, and starts our UDP listeners
+	if err = peerConnection.SetLocalDescription(answer); err != nil {
+		panic(err)
+	}
+
+	// Block until ICE Gathering is complete, disabling trickle ICE
+	// we do this because we only can exchange one signaling message
+	// in a production application you should exchange ICE Candidates via OnICECandidate
+	<-gatherComplete
+
+	// Output the answer
+	localSdpChan <- *peerConnection.LocalDescription()
+
+	// Block forever
+	select {}
+}
+
+// Search for Codec PayloadType
+//
+// Since we are answering we need to match the remote PayloadType
+func getPayloadType(m webrtc.MediaEngine, codecType webrtc.RTPCodecType, codecName string) uint8 {
+	for _, codec := range m.GetCodecsByKind(codecType) {
+		if codec.Name == codecName {
+			return codec.PayloadType
+		}
+	}
+	panic(fmt.Sprintf("Remote peer does not support %s", codecName))
+}

web/static/css/style.css

@@ -76,6 +76,9 @@ h1, h2, h3, h4 {
   width: 100%;
   height: 100%;
   max-height: 90vh;
+
+  /* Black borders when video is not 16/9 */
+  background-color: #000;
 }
 
 .col-chat {

web/static/js/viewer.js

@@ -12,7 +12,7 @@ peerConnection = new RTCPeerConnection({
 peerConnection.oniceconnectionstatechange = e => {
     console.log(peerConnection.iceConnectionState)
 
-    switch (myPeerConnection.iceConnectionState) {
+    switch (peerConnection.iceConnectionState) {
         case "closed":
         case "failed":
             console.log("FIXME Failed");

web/web.go

@@ -20,11 +20,19 @@ type Options struct {
 	WidgetURL     string
 }
 
-// Preload templates
+var (
-var templates = template.Must(template.ParseGlob("web/template/*.html"))
+	cfg *Options
+
+	// WebRTC session description channels
+	remoteSdpChan chan webrtc.SessionDescription
+	localSdpChan  chan webrtc.SessionDescription
+
+	// Preload templates
+	templates = template.Must(template.ParseGlob("web/template/*.html"))
+)
 
 // Handle WebRTC session description exchange via POST
-func sessionExchangeHandler(w http.ResponseWriter, r *http.Request) {
+func viewerPostHandler(w http.ResponseWriter, r *http.Request) {
 	// Limit response body to 128KB
 	r.Body = http.MaxBytesReader(w, r.Body, 131072)
 
@@ -37,8 +45,9 @@ func sessionExchangeHandler(w http.ResponseWriter, r *http.Request) {
 		return
 	}
 
-	// FIXME remoteDescription -> "Magic" -> localDescription
+	// Exchange session descriptions with WebRTC stream server
-	localDescription := remoteDescription
+	remoteSdpChan <- remoteDescription
+	localDescription := <-localSdpChan
 
 	// Send server description as JSON
 	jsonDesc, err := json.Marshal(localDescription)
@@ -49,15 +58,12 @@ func sessionExchangeHandler(w http.ResponseWriter, r *http.Request) {
 	}
 	w.Header().Set("Content-Type", "application/json")
 	w.Write(jsonDesc)
+
+	// Increment monitoring
+	monitoring.WebSessions.Inc()
 }
 
-// Handle site index and viewer pages
+func viewerGetHandler(w http.ResponseWriter, r *http.Request) {
-// POST requests are used to exchange WebRTC session descriptions
-func viewerHandler(w http.ResponseWriter, r *http.Request, cfg *Options) {
-	// FIXME validation on path: https://golang.org/doc/articles/wiki/#tmp_11
-
-	switch r.Method {
-	case "GET":
 	// Render template
 	data := struct {
 		Path string
@@ -66,20 +72,32 @@ func viewerHandler(w http.ResponseWriter, r *http.Request, cfg *Options) {
 	if err := templates.ExecuteTemplate(w, "base", data); err != nil {
 		log.Println(err.Error())
 		http.Error(w, "Internal Server Error", http.StatusInternalServerError)
-		}
+		return
-	case "POST":
-		sessionExchangeHandler(w, r)
-	default:
-		http.Error(w, "Sorry, only GET and POST methods are supported.", http.StatusBadRequest)
 	}
 
 	// Increment monitoring
-	monitoring.ViewerServed.Inc()
+	monitoring.WebViewerServed.Inc()
+}
+
+// Handle site index and viewer pages
+// POST requests are used to exchange WebRTC session descriptions
+func viewerHandler(w http.ResponseWriter, r *http.Request) {
+	// FIXME validation on path: https://golang.org/doc/articles/wiki/#tmp_11
+
+	// Route depending on HTTP method
+	switch r.Method {
+	case http.MethodGet:
+		viewerGetHandler(w, r)
+	case http.MethodPost:
+		viewerPostHandler(w, r)
+	default:
+		http.Error(w, "Sorry, only GET and POST methods are supported.", http.StatusBadRequest)
+	}
 }
 
 // Handle static files
 // We do not use http.FileServer as we do not want directory listing
-func staticHandler(w http.ResponseWriter, r *http.Request, cfg *Options) {
+func staticHandler(w http.ResponseWriter, r *http.Request) {
 	path := "./web/" + r.URL.Path
 	if f, err := os.Stat(path); err == nil && !f.IsDir() {
 		http.ServeFile(w, r, path)
@@ -88,19 +106,16 @@ func staticHandler(w http.ResponseWriter, r *http.Request, cfg *Options) {
 	}
 }
 
-// Closure to pass configuration
+// Serve HTTP server
-func makeHandler(fn func(http.ResponseWriter, *http.Request, *Options), cfg *Options) http.HandlerFunc {
+func Serve(rSdpChan chan webrtc.SessionDescription, lSdpChan chan webrtc.SessionDescription, c *Options) {
-	return func(w http.ResponseWriter, r *http.Request) {
+	remoteSdpChan = rSdpChan
-		fn(w, r, cfg)
+	localSdpChan = lSdpChan
-	}
+	cfg = c
-}
 
-// ServeHTTP server
-func ServeHTTP(cfg *Options) {
 	// Set up HTTP router and server
 	mux := http.NewServeMux()
-	mux.HandleFunc("/", makeHandler(viewerHandler, cfg))
+	mux.HandleFunc("/", viewerHandler)
-	mux.HandleFunc("/static/", makeHandler(staticHandler, cfg))
+	mux.HandleFunc("/static/", staticHandler)
 	log.Printf("HTTP server listening on %s", cfg.ListenAddress)
 	log.Fatal(http.ListenAndServe(cfg.ListenAddress, mux))
 }