Test WebRTC Serve

stream/webrtc/webrtc.go

@@ -2,7 +2,6 @@
 package webrtc
 
 import (
-	"fmt"
 	"log"
 	"math/rand"
 	"strings"
@@ -166,7 +165,7 @@ func newPeerHandler(localSdpChan chan webrtc.SessionDescription, remoteSdp struc
 	<-gatherComplete
 
 	// Send answer to client
-	localSdpChan <- answer
+	localSdpChan <- *peerConnection.LocalDescription()
 }
 
 // Search for Codec PayloadType
@@ -178,7 +177,8 @@ func getPayloadType(m webrtc.MediaEngine, codecType webrtc.RTPCodecType, codecNa
 			return codec, codec.PayloadType
 		}
 	}
-	panic(fmt.Sprintf("Remote peer does not support %s", codecName))
+	log.Printf("Remote peer does not support %s", codecName)
+	return nil, 0
 }
 
 // Serve WebRTC media streaming server

stream/webrtc/webrtc_test.go

@@ -1 +1,69 @@
 package webrtc
+
+import (
+	"math/rand"
+	"testing"
+
+	"github.com/pion/webrtc/v3"
+	"gitlab.crans.org/nounous/ghostream/stream/srt"
+)
+
+func TestServe(t *testing.T) {
+	// Serve WebRTC server
+	remoteSdpChan := make(chan struct {
+		StreamID          string
+		RemoteDescription webrtc.SessionDescription
+	})
+	localSdpChan := make(chan webrtc.SessionDescription)
+	webrtcChannel := make(chan srt.Packet, 64)
+	cfg := Options{
+		Enabled:     true,
+		MinPortUDP:  10000,
+		MaxPortUDP:  10005,
+		STUNServers: []string{"stun:stun.l.google.com:19302"},
+	}
+	go Serve(remoteSdpChan, localSdpChan, webrtcChannel, &cfg)
+
+	// New client connection
+	mediaEngine := webrtc.MediaEngine{}
+	mediaEngine.RegisterDefaultCodecs()
+	api := webrtc.NewAPI(webrtc.WithMediaEngine(mediaEngine))
+	peerConnection, _ := api.NewPeerConnection(webrtc.Configuration{})
+
+	// Create video track
+	codec, payloadType := getPayloadType(mediaEngine, webrtc.RTPCodecTypeVideo, "VP8")
+	videoTrack, err := webrtc.NewTrack(payloadType, rand.Uint32(), "video", "pion", codec)
+	if err != nil {
+		t.Error("Failed to create new video track", err)
+	}
+	if _, err = peerConnection.AddTrack(videoTrack); err != nil {
+		t.Error("Failed to add video track", err)
+	}
+
+	// Create audio track
+	codec, payloadType = getPayloadType(mediaEngine, webrtc.RTPCodecTypeAudio, "opus")
+	audioTrack, err := webrtc.NewTrack(payloadType, rand.Uint32(), "audio", "pion", codec)
+	if err != nil {
+		t.Error("Failed to create new audio track", err)
+	}
+	if _, err = peerConnection.AddTrack(audioTrack); err != nil {
+		t.Error("Failed to add audio track", err)
+	}
+
+	// Create offer
+	offer, _ := peerConnection.CreateOffer(nil)
+
+	// Create channel that is blocked until ICE Gathering is complete
+	gatherComplete := webrtc.GatheringCompletePromise(peerConnection)
+	peerConnection.SetLocalDescription(offer)
+	<-gatherComplete
+
+	// Send offer to server
+	remoteSdpChan <- struct {
+		StreamID          string
+		RemoteDescription webrtc.SessionDescription
+	}{"demo", *peerConnection.LocalDescription()}
+	_ = <-localSdpChan
+
+	// FIXME: verify connection did work
+}