Properly handle non-media probes

interceptor_test.go

@@ -225,12 +225,59 @@ func Test_InterceptorRegistry_Build(t *testing.T) {
 		},
 	})
 
-	peerConnectionA, err := NewAPI(WithInterceptorRegistry(ir)).NewPeerConnection(Configuration{})
-	assert.NoError(t, err)
-
-	peerConnectionB, err := NewAPI(WithInterceptorRegistry(ir)).NewPeerConnection(Configuration{})
+	peerConnectionA, peerConnectionB, err := NewAPI(WithInterceptorRegistry(ir)).newPair(Configuration{})
 	assert.NoError(t, err)
 
 	assert.Equal(t, 2, registryBuildCount)
 	closePairNow(t, peerConnectionA, peerConnectionB)
 }
+
+func Test_Interceptor_ZeroSSRC(t *testing.T) {
+	to := test.TimeOut(time.Second * 20)
+	defer to.Stop()
+
+	report := test.CheckRoutines(t)
+	defer report()
+
+	track, err := NewTrackLocalStaticRTP(RTPCodecCapability{MimeType: MimeTypeVP8}, "video", "pion")
+	assert.NoError(t, err)
+
+	offerer, answerer, err := newPair()
+	assert.NoError(t, err)
+
+	_, err = offerer.AddTrack(track)
+	assert.NoError(t, err)
+
+	probeReceiverCreated := make(chan struct{})
+
+	go func() {
+		sequenceNumber := uint16(0)
+		for range time.NewTicker(time.Millisecond * 20).C {
+			track.mu.Lock()
+			if len(track.bindings) == 1 {
+				_, err = track.bindings[0].writeStream.WriteRTP(&rtp.Header{
+					Version:        2,
+					SSRC:           0,
+					SequenceNumber: sequenceNumber,
+				}, []byte{0, 1, 2, 3, 4, 5})
+				assert.NoError(t, err)
+			}
+			sequenceNumber++
+			track.mu.Unlock()
+
+			if nonMediaBandwidthProbe, ok := answerer.nonMediaBandwidthProbe.Load().(*RTPReceiver); ok {
+				assert.Equal(t, len(nonMediaBandwidthProbe.Tracks()), 1)
+				close(probeReceiverCreated)
+				return
+			}
+		}
+	}()
+
+	assert.NoError(t, signalPair(offerer, answerer))
+
+	peerConnectionConnected := untilConnectionState(PeerConnectionStateConnected, offerer, answerer)
+	peerConnectionConnected.Wait()
+
+	<-probeReceiverCreated
+	closePairNow(t, offerer, answerer)
+}

peerconnection.go

@@ -68,7 +68,8 @@
 	// should be defined (see JSEP 3.4.1).
 	greaterMid int
 
-	rtpTransceivers []*RTPTransceiver
+	rtpTransceivers        []*RTPTransceiver
+	nonMediaBandwidthProbe atomic.Value // RTPReceiver
 
 	onSignalingStateChangeHandler     func(SignalingState)
 	onICEConnectionStateChangeHandler atomic.Value // func(ICEConnectionState)
@@ -1524,6 +1525,32 @@
 	return true, nil
 }
 
+// Chrome sends probing traffic on SSRC 0. This reads the packets to ensure that we properly
+// generate TWCC reports for it. Since this isn't actually media we don't pass this to the user
+func (pc *PeerConnection) handleNonMediaBandwidthProbe() {
+	nonMediaBandwidthProbe, err := pc.api.NewRTPReceiver(RTPCodecTypeVideo, pc.dtlsTransport)
+	if err != nil {
+		pc.log.Errorf("handleNonMediaBandwidthProbe failed to create RTPReceiver: %v", err)
+		return
+	}
+
+	if err = nonMediaBandwidthProbe.Receive(RTPReceiveParameters{
+		Encodings: []RTPDecodingParameters{{RTPCodingParameters: RTPCodingParameters{}}},
+	}); err != nil {
+		pc.log.Errorf("handleNonMediaBandwidthProbe failed to start RTPReceiver: %v", err)
+		return
+	}
+
+	pc.nonMediaBandwidthProbe.Store(nonMediaBandwidthProbe)
+	b := make([]byte, pc.api.settingEngine.getReceiveMTU())
+	for {
+		if _, _, err = nonMediaBandwidthProbe.readRTP(b, nonMediaBandwidthProbe.Track()); err != nil {
+			pc.log.Tracef("handleNonMediaBandwidthProbe read exiting: %v", err)
+			return
+		}
+	}
+}
+
 func (pc *PeerConnection) handleIncomingSSRC(rtpStream io.Reader, ssrc SSRC) error { //nolint:gocognit
 	remoteDescription := pc.RemoteDescription()
 	if remoteDescription == nil {
@@ -1656,6 +1683,11 @@
 			continue
 		}
 
+		if ssrc == 0 {
+			go pc.handleNonMediaBandwidthProbe()
+			continue
+		}
+
 		pc.dtlsTransport.storeSimulcastStream(stream)
 
 		if atomic.AddUint64(&simulcastRoutineCount, 1) >= simulcastMaxProbeRoutines {
@@ -2072,6 +2104,9 @@
 			closeErrs = append(closeErrs, t.Stop())
 		}
 	}
+	if nonMediaBandwidthProbe, ok := pc.nonMediaBandwidthProbe.Load().(*RTPReceiver); ok {
+		closeErrs = append(closeErrs, nonMediaBandwidthProbe.Stop())
+	}
 	pc.mu.Unlock()
 
 	// https://www.w3.org/TR/webrtc/#dom-rtcpeerconnection-close (step #5)

peerconnection_media_test.go

@@ -1528,7 +1528,7 @@ func TestPeerConnection_Simulcast_RTX(t *testing.T) {
 					SequenceNumber: sequenceNumber,
 					PayloadType:    96,
 					Padding:        true,
-					SSRC:           uint32(i),
+					SSRC:           uint32(i + 1),
 				},
 				Payload: []byte{0x00, 0x02},
 			}
@@ -1547,7 +1547,7 @@ func TestPeerConnection_Simulcast_RTX(t *testing.T) {
 					Version:        2,
 					SequenceNumber: sequenceNumber,
 					PayloadType:    96,
-					SSRC:           uint32(i),
+					SSRC:           uint32(i + 1),
 				},
 				Payload: []byte{0x00},
 			}
@@ -1591,7 +1591,7 @@ func TestPeerConnection_Simulcast_RTX(t *testing.T) {
 					Version:        2,
 					SequenceNumber: sequenceNumber,
 					PayloadType:    96,
-					SSRC:           uint32(i),
+					SSRC:           uint32(i + 1),
 				},
 				Payload: []byte{0x00},
 			}

peerconnection_renegotiation_test.go

@@ -1046,7 +1046,7 @@ func TestPeerConnection_Renegotiation_Simulcast(t *testing.T) {
 			for ssrc, rid := range rids {
 				header := &rtp.Header{
 					Version:        2,
-					SSRC:           uint32(ssrc),
+					SSRC:           uint32(ssrc + 1),
 					SequenceNumber: sequenceNumber,
 					PayloadType:    96,
 				}

rtpreceiver.go

@@ -201,7 +201,7 @@
 
 		var t *trackStreams
 		for idx, ts := range r.tracks {
-			if ts.track != nil && parameters.Encodings[i].SSRC != 0 && ts.track.SSRC() == parameters.Encodings[i].SSRC {
+			if ts.track != nil && ts.track.SSRC() == parameters.Encodings[i].SSRC {
 				t = &r.tracks[idx]
 				break
 			}
@@ -210,12 +210,10 @@
 			return fmt.Errorf("%w: %d", errRTPReceiverWithSSRCTrackStreamNotFound, parameters.Encodings[i].SSRC)
 		}
 
-		if parameters.Encodings[i].SSRC != 0 {
-			t.streamInfo = createStreamInfo("", parameters.Encodings[i].SSRC, 0, codec, globalParams.HeaderExtensions)
-			var err error
-			if t.rtpReadStream, t.rtpInterceptor, t.rtcpReadStream, t.rtcpInterceptor, err = r.transport.streamsForSSRC(parameters.Encodings[i].SSRC, *t.streamInfo); err != nil {
-				return err
-			}
+		t.streamInfo = createStreamInfo("", parameters.Encodings[i].SSRC, 0, codec, globalParams.HeaderExtensions)
+		var err error
+		if t.rtpReadStream, t.rtpInterceptor, t.rtcpReadStream, t.rtcpInterceptor, err = r.transport.streamsForSSRC(parameters.Encodings[i].SSRC, *t.streamInfo); err != nil {
+			return err
 		}
 
 		if rtxSsrc := parameters.Encodings[i].RTX.SSRC; rtxSsrc != 0 {