diff --git a/dtlstransport.go b/dtlstransport.go
index ec08a0846d3..bfc38d2b724 100644
--- a/dtlstransport.go
+++ b/dtlstransport.go
@@ -51,7 +51,7 @@ type DTLSTransport struct {
 
 	srtpSession, srtcpSession   atomic.Value
 	srtpEndpoint, srtcpEndpoint *mux.Endpoint
-	simulcastStreams            []*srtp.ReadStreamSRTP
+	simulcastStreams            []simulcastStreamPair
 	srtpReady                   chan struct{}
 
 	dtlsMatcher mux.MatchFunc
@@ -60,6 +60,11 @@ type DTLSTransport struct {
 	log logging.LeveledLogger
 }
 
+type simulcastStreamPair struct {
+	srtp  *srtp.ReadStreamSRTP
+	srtcp *srtp.ReadStreamSRTCP
+}
+
 // NewDTLSTransport creates a new DTLSTransport.
 // This constructor is part of the ORTC API. It is not
 // meant to be used together with the basic WebRTC API.
@@ -436,7 +441,8 @@ func (t *DTLSTransport) Stop() error {
 	}
 
 	for i := range t.simulcastStreams {
-		closeErrs = append(closeErrs, t.simulcastStreams[i].Close())
+		closeErrs = append(closeErrs, t.simulcastStreams[i].srtp.Close())
+		closeErrs = append(closeErrs, t.simulcastStreams[i].srtcp.Close())
 	}
 
 	if t.conn != nil {
@@ -477,11 +483,11 @@ func (t *DTLSTransport) ensureICEConn() error {
 	return nil
 }
 
-func (t *DTLSTransport) storeSimulcastStream(s *srtp.ReadStreamSRTP) {
+func (t *DTLSTransport) storeSimulcastStream(srtpReadStream *srtp.ReadStreamSRTP, srtcpReadStream *srtp.ReadStreamSRTCP) {
 	t.lock.Lock()
 	defer t.lock.Unlock()
 
-	t.simulcastStreams = append(t.simulcastStreams, s)
+	t.simulcastStreams = append(t.simulcastStreams, simulcastStreamPair{srtpReadStream, srtcpReadStream})
 }
 
 func (t *DTLSTransport) streamsForSSRC(ssrc SSRC, streamInfo interceptor.StreamInfo) (*srtp.ReadStreamSRTP, interceptor.RTPReader, *srtp.ReadStreamSRTCP, interceptor.RTCPReader, error) {
diff --git a/peerconnection.go b/peerconnection.go
index 5876e5f76cd..2b8a9038539 100644
--- a/peerconnection.go
+++ b/peerconnection.go
@@ -1670,26 +1670,42 @@ func (pc *PeerConnection) undeclaredRTPMediaProcessor() {
 			return
 		}
 
-		stream, ssrc, err := srtpSession.AcceptStream()
+		srtcpSession, err := pc.dtlsTransport.getSRTCPSession()
+		if err != nil {
+			pc.log.Warnf("undeclaredMediaProcessor failed to open SrtcpSession: %v", err)
+			return
+		}
+
+		srtpReadStream, ssrc, err := srtpSession.AcceptStream()
 		if err != nil {
 			pc.log.Warnf("Failed to accept RTP %v", err)
 			return
 		}
 
+		// open accompanying srtcp stream
+		srtcpReadStream, err := srtcpSession.OpenReadStream(ssrc)
+		if err != nil {
+			pc.log.Warnf("Failed to open RTCP stream for %d: %v", ssrc, err)
+			return
+		}
+
 		if pc.isClosed.get() {
-			if err = stream.Close(); err != nil {
+			if err = srtpReadStream.Close(); err != nil {
 				pc.log.Warnf("Failed to close RTP stream %v", err)
 			}
+			if err = srtcpReadStream.Close(); err != nil {
+				pc.log.Warnf("Failed to close RTCP stream %v", err)
+			}
 			continue
 		}
 
+		pc.dtlsTransport.storeSimulcastStream(srtpReadStream, srtcpReadStream)
+
 		if ssrc == 0 {
 			go pc.handleNonMediaBandwidthProbe()
 			continue
 		}
 
-		pc.dtlsTransport.storeSimulcastStream(stream)
-
 		if atomic.AddUint64(&simulcastRoutineCount, 1) >= simulcastMaxProbeRoutines {
 			atomic.AddUint64(&simulcastRoutineCount, ^uint64(0))
 			pc.log.Warn(ErrSimulcastProbeOverflow.Error())
@@ -1701,7 +1717,7 @@ func (pc *PeerConnection) undeclaredRTPMediaProcessor() {
 				pc.log.Errorf(incomingUnhandledRTPSsrc, ssrc, err)
 			}
 			atomic.AddUint64(&simulcastRoutineCount, ^uint64(0))
-		}(stream, SSRC(ssrc))
+		}(srtpReadStream, SSRC(ssrc))
 	}
 }