private static async Task StartFfmpegListener(string sdpPath, CancellationToken cancel)
{
while (!File.Exists(FFMPEG_SDP_FILE) && !cancel.IsCancellationRequested)
{
await Task.Delay(500);
}
if (!cancel.IsCancellationRequested)
{
var sdp = SDP.ParseSDPDescription(File.ReadAllText(FFMPEG_SDP_FILE));
// The SDP is only expected to contain a single video media announcement.
var videoAnn = sdp.Media.Single(x => x.Media == SDPMediaTypesEnum.video);
_ffmpegVideoFormat = videoAnn.MediaFormats.Values.First();
_ffmpegListener = new RTPSession(false, false, false, IPAddress.Loopback, FFMPEG_DEFAULT_RTP_PORT);
_ffmpegListener.AcceptRtpFromAny = true;
MediaStreamTrack videoTrack = new MediaStreamTrack(SDPMediaTypesEnum.video, false, new List <SDPAudioVideoMediaFormat> {
_ffmpegVideoFormat
}, MediaStreamStatusEnum.RecvOnly);
_ffmpegListener.addTrack(videoTrack);
_ffmpegListener.SetRemoteDescription(SIP.App.SdpType.answer, sdp);
// Set a dummy destination end point or the RTP session will end up sending RTCP reports
// to itself.
var dummyIPEndPoint = new IPEndPoint(IPAddress.Loopback, 0);
_ffmpegListener.SetDestination(SDPMediaTypesEnum.video, dummyIPEndPoint, dummyIPEndPoint);
await _ffmpegListener.Start();
}
}
public void MapWellKnownAudioFormatUnitTest()
{
logger.LogDebug("--> " + System.Reflection.MethodBase.GetCurrentMethod().Name);
logger.BeginScope(System.Reflection.MethodBase.GetCurrentMethod().Name);
SDPAudioVideoMediaFormat pcmu = new SDPAudioVideoMediaFormat(SDPWellKnownMediaFormatsEnum.PCMU);
var audioFormat = pcmu.ToAudioFormat();
Assert.Equal(SDPMediaTypesEnum.audio, pcmu.Kind);
Assert.Equal(AudioCodecsEnum.PCMU, audioFormat.Codec);
Assert.Equal(8000, audioFormat.ClockRate);
Assert.Equal("PCMU/8000", pcmu.Rtpmap);
}
private static RTCPeerConnection Createpc(WebSocketContext context, SDPAudioVideoMediaFormat videoFormat)
{
var pc = new RTCPeerConnection(null);
MediaStreamTrack videoTrack = new MediaStreamTrack(SDPMediaTypesEnum.video, false, new List <SDPAudioVideoMediaFormat> {
videoFormat
}, MediaStreamStatusEnum.SendOnly);
pc.addTrack(videoTrack);
pc.onicecandidateerror += (candidate, error) => logger.LogWarning($"Error adding remote ICE candidate. {error} {candidate}");
pc.oniceconnectionstatechange += (state) => logger.LogDebug($"ICE connection state change to {state}.");
//pc.OnReceiveReport += (type, rtcp) => logger.LogDebug($"RTCP {type} report received.");
pc.OnRtcpBye += (reason) => logger.LogDebug($"RTCP BYE receive, reason: {(string.IsNullOrWhiteSpace(reason) ? "<none>" : reason)}.");
pc.OnRtpClosed += (reason) => logger.LogDebug($"Peer connection closed, reason: {(string.IsNullOrWhiteSpace(reason) ? "<none>" : reason)}.");
pc.onicecandidate += (candidate) =>
{
if (pc.signalingState == RTCSignalingState.have_local_offer ||
pc.signalingState == RTCSignalingState.have_remote_offer)
{
context.WebSocket.Send($"candidate:{candidate}");
}
};
pc.onconnectionstatechange += (state) =>
{
logger.LogDebug($"Peer connection state changed to {state}.");
if (state == RTCPeerConnectionState.connected)
{
logger.LogDebug("Creating RTP session to receive ffmpeg stream.");
_ffmpegListener.OnRtpPacketReceived += (ep, media, rtpPkt) =>
{
if (media == SDPMediaTypesEnum.video && pc.VideoDestinationEndPoint != null)
{
//logger.LogDebug($"Forwarding {media} RTP packet to webrtc peer timestamp {rtpPkt.Header.Timestamp}.");
pc.SendRtpRaw(media, rtpPkt.Payload, rtpPkt.Header.Timestamp, rtpPkt.Header.MarkerBit, rtpPkt.Header.PayloadType);
}
};
}
};
return(pc);
}
