webrtc: fix error "Failed to setup RTCP mux" on some readers (#3381) (#3449)

internal/protocols/webrtc/peer_connection.go

@@ -128,6 +128,9 @@ func (co *PeerConnection) Start() error {
 	mediaEngine := &webrtc.MediaEngine{}
 
 	if co.Publish {
+		videoSetupped := false
+		audioSetupped := false
+
 		for _, tr := range co.OutgoingTracks {
 			params, err := tr.codecParameters()
 			if err != nil {
@@ -137,8 +140,10 @@ func (co *PeerConnection) Start() error {
 			var codecType webrtc.RTPCodecType
 			if tr.isVideo() {
 				codecType = webrtc.RTPCodecTypeVideo
+				videoSetupped = true
 			} else {
 				codecType = webrtc.RTPCodecTypeAudio
+				audioSetupped = true
 			}
 
 			err = mediaEngine.RegisterCodec(params, codecType)
@@ -146,6 +151,33 @@ func (co *PeerConnection) Start() error {
 				return err
 			}
 		}
+
+		// always register at least a video and a audio codec
+		// otherwise handshake will fail or audio will be muted on some clients (like Firefox)
+		if !videoSetupped {
+			err := mediaEngine.RegisterCodec(webrtc.RTPCodecParameters{
+				RTPCodecCapability: webrtc.RTPCodecCapability{
+					MimeType:  webrtc.MimeTypeVP8,
+					ClockRate: 90000,
+				},
+				PayloadType: 96,
+			}, webrtc.RTPCodecTypeVideo)
+			if err != nil {
+				return err
+			}
+		}
+		if !audioSetupped {
+			err := mediaEngine.RegisterCodec(webrtc.RTPCodecParameters{
+				RTPCodecCapability: webrtc.RTPCodecCapability{
+					MimeType:  webrtc.MimeTypePCMU,
+					ClockRate: 8000,
+				},
+				PayloadType: 0,
+			}, webrtc.RTPCodecTypeAudio)
+			if err != nil {
+				return err
+			}
+		}
 	} else {
 		for _, codec := range incomingVideoCodecs {
 			err := mediaEngine.RegisterCodec(codec, webrtc.RTPCodecTypeVideo)

internal/protocols/webrtc/peer_connection_test.go

@@ -9,6 +9,7 @@ import (
 	"github.com/bluenviron/mediamtx/internal/conf"
 	"github.com/bluenviron/mediamtx/internal/test"
 	"github.com/pion/rtp"
+	"github.com/pion/sdp/v3"
 	"github.com/pion/webrtc/v3"
 	"github.com/stretchr/testify/require"
 )
@@ -409,3 +410,68 @@ func TestPeerConnectionPublishRead(t *testing.T) {
 		})
 	}
 }
+
+// test that an audio codec is present regardless of the fact that an audio track is not.
+func TestPeerConnectionFallbackCodecs(t *testing.T) {
+	pc1 := &PeerConnection{
+		HandshakeTimeout:   conf.StringDuration(10 * time.Second),
+		TrackGatherTimeout: conf.StringDuration(2 * time.Second),
+		LocalRandomUDP:     true,
+		IPsFromInterfaces:  true,
+		Publish:            false,
+		Log:                test.NilLogger,
+	}
+	err := pc1.Start()
+	require.NoError(t, err)
+	defer pc1.Close()
+
+	pc2 := &PeerConnection{
+		HandshakeTimeout:   conf.StringDuration(10 * time.Second),
+		TrackGatherTimeout: conf.StringDuration(2 * time.Second),
+		LocalRandomUDP:     true,
+		IPsFromInterfaces:  true,
+		Publish:            true,
+		OutgoingTracks: []*OutgoingTrack{{
+			Format: &format.AV1{
+				PayloadTyp: 96,
+			},
+		}},
+		Log: test.NilLogger,
+	}
+	err = pc2.Start()
+	require.NoError(t, err)
+	defer pc2.Close()
+
+	offer, err := pc1.CreatePartialOffer()
+	require.NoError(t, err)
+
+	answer, err := pc2.CreateFullAnswer(context.Background(), offer)
+	require.NoError(t, err)
+
+	var s sdp.SessionDescription
+	err = s.Unmarshal([]byte(answer.SDP))
+	require.NoError(t, err)
+
+	require.Equal(t, []*sdp.MediaDescription{
+		{
+			MediaName: sdp.MediaName{
+				Media:   "video",
+				Port:    sdp.RangedPort{Value: 9},
+				Protos:  []string{"UDP", "TLS", "RTP", "SAVPF"},
+				Formats: []string{"97"},
+			},
+			ConnectionInformation: s.MediaDescriptions[0].ConnectionInformation,
+			Attributes:            s.MediaDescriptions[0].Attributes,
+		},
+		{
+			MediaName: sdp.MediaName{
+				Media:   "audio",
+				Port:    sdp.RangedPort{Value: 9},
+				Protos:  []string{"UDP", "TLS", "RTP", "SAVPF"},
+				Formats: []string{"0"},
+			},
+			ConnectionInformation: s.MediaDescriptions[1].ConnectionInformation,
+			Attributes:            s.MediaDescriptions[1].Attributes,
+		},
+	}, s.MediaDescriptions)
+}