Added missing event handler for RTCP report ready. (sipsorcery-org#1269)

Author: sipsorcery

examples/WebRTCExamples/WebRTCReceiveAudio/Program.cs

@@ -119,7 +119,6 @@ private static async Task<RTCPeerConnection> SendSDPOffer(WebSocketContext conte
             peerConnection.OnAudioFormatsNegotiated += (audioFormats) =>
                 windowsAudioEP.SetAudioSinkFormat(audioFormats.First());
             peerConnection.OnReceiveReport += RtpSession_OnReceiveReport;
-            peerConnection.OnSendReport += RtpSession_OnSendReport;
             peerConnection.OnSendReportByIndex += RtpSession_OnSendReportByIndex;
             peerConnection.OnTimeout += (mediaType) => logger.LogDebug($"Timeout on media {mediaType}.");
             peerConnection.oniceconnectionstatechange += (state) => logger.LogDebug($"ICE connection state changed to {state}.");
@@ -134,7 +133,7 @@ private static async Task<RTCPeerConnection> SendSDPOffer(WebSocketContext conte
                 else if (state == RTCPeerConnectionState.closed || state == RTCPeerConnectionState.failed)
                 {
                     peerConnection.OnReceiveReport -= RtpSession_OnReceiveReport;
-                    peerConnection.OnSendReport -= RtpSession_OnSendReport;
+                    peerConnection.OnSendReportByIndex -= RtpSession_OnSendReportByIndex;
 
                     await windowsAudioEP.CloseAudio();
                 }
@@ -182,15 +181,12 @@ private static void WebSocketMessageReceived(RTCPeerConnection peerConnection, s
             }
         }
 
-        private static void RtpSession_OnSendReport(SDPMediaTypesEnum mediaType, RTCPCompoundPacket sentRtcpReport)
-            => RtpSession_OnSendReportByIndex(0, mediaType, sentRtcpReport);
-
         /// <summary>
         /// Diagnostic handler to print out our RTCP sender/receiver reports.
         /// </summary>
         private static void RtpSession_OnSendReportByIndex(int index, SDPMediaTypesEnum mediaType, RTCPCompoundPacket sentRtcpReport)
         {
-            logger.LogDebug($"RTCP report sent for index {index} and media {mediaType}.");
+            //logger.LogDebug($"RTCP report sent for index {index} and media {mediaType}.");
 
             if (sentRtcpReport.Bye != null)
             {
@@ -206,7 +202,7 @@ private static void RtpSession_OnSendReportByIndex(int index, SDPMediaTypesEnum
                 if (sentRtcpReport.ReceiverReport.ReceptionReports?.Count > 0)
                 {
                     var rrSample = sentRtcpReport.ReceiverReport.ReceptionReports.First();
-                    logger.LogDebug($"RTCP sent RR {mediaType}, ssrc {rrSample.SSRC}, seqnum {rrSample.ExtendedHighestSequenceNumber}.");
+                    logger.LogDebug($"RTCP sent RR {mediaType}, ssrc {rrSample.SSRC}, seqnum {rrSample.ExtendedHighestSequenceNumber}, pkts lost {rrSample.PacketsLost}.");
                 }
                 else
                 {
@@ -220,22 +216,22 @@ private static void RtpSession_OnSendReportByIndex(int index, SDPMediaTypesEnum
         /// </summary>
         private static void RtpSession_OnReceiveReport(IPEndPoint remoteEndPoint, SDPMediaTypesEnum mediaType, RTCPCompoundPacket recvRtcpReport)
         {
-            logger.LogDebug($"RTCP report received for {mediaType}.");
+            //logger.LogDebug($"RTCP report received for {mediaType}.");
 
             if (recvRtcpReport.Bye != null)
             {
                 logger.LogDebug($"RTCP recv BYE {mediaType}.");
             }
-            else
+            else if(recvRtcpReport.ReceiverReport != null)
             {
-                var rr = recvRtcpReport.ReceiverReport?.ReceptionReports?.FirstOrDefault();
+                var rr = recvRtcpReport.ReceiverReport.ReceptionReports?.FirstOrDefault();
                 if (rr != null)
                 {
-                    logger.LogDebug($"RTCP {mediaType} Receiver Report: SSRC {rr.SSRC}, pkts lost {rr.PacketsLost}, delay since SR {rr.DelaySinceLastSenderReport}.");
+                    logger.LogDebug($"RTCP {mediaType} Receiver Report SSRC {rr.SSRC}: pkts lost {rr.PacketsLost}, delay since SR {rr.DelaySinceLastSenderReport}.");
                 }
                 else
                 {
-                    logger.LogDebug($"RTCP {mediaType} Receiver Report: empty.");
+                    logger.LogDebug($"RTCP {mediaType} Receiver Report for SSRC {recvRtcpReport.ReceiverReport.SSRC}: empty.");
                 }
             }
         }

src/net/RTP/MediaStream.cs

@@ -673,7 +673,6 @@ public void OnReceiveRTPPacket(RTPHeader hdr, int localPort, IPEndPoint remoteEn
                 }
             }
 
-
             // Note AC 24 Dec 2020: The problem with waiting until the remote description is set is that the remote peer often starts sending
             // RTP packets at the same time it signals its SDP offer or answer. Generally this is not a problem for audio but for video streams
             // the first RTP packet(s) are the key frame and if they are ignored the video stream will take additional time or manual 

src/net/RTP/RTPSession.cs

@@ -655,6 +655,7 @@ private void CreateRtcpSession(MediaStream mediaStream)
                 mediaStream.OnRtpPacketReceivedByIndex += RaisedOnRtpPacketReceived;
                 mediaStream.OnRtpHeaderReceivedByIndex += RaisedOnRtpHeaderReceived;
                 mediaStream.OnReceiveReportByIndex += RaisedOnOnReceiveReport;
+                mediaStream.RtcpSession.OnReportReadyToSend += SendRtcpReport;
 
                 if (mediaStream.MediaType == SDPMediaTypesEnum.audio)
                 {