/// <summary>
/// Sends two separate RTP streams to an application like ffplay.
///
/// ffplay -i ffplay_av.sdp -protocol_whitelist "file,rtp,udp" -loglevel debug
///
/// The SDP that describes the streams is:
///
/// v=0
/// o=- 1129870806 2 IN IP4 127.0.0.1
/// s=-
/// c=IN IP4 192.168.11.50
/// t=0 0
/// m=audio 4040 RTP/AVP 0
/// a=rtpmap:0 PCMU/8000
/// m=video 4042 RTP/AVP 100
/// a=rtpmap:100 VP8/90000
/// </summary>
private void SendSamplesAsRtp(IPEndPoint dstBaseEndPoint)
{
try
{
Socket videoSrcRtpSocket = null;
Socket videoSrcControlSocket = null;
Socket audioSrcRtpSocket = null;
Socket audioSrcControlSocket = null;
// WebRtc multiplexes all the RTP and RTCP sessions onto a single UDP connection.
// The approach needed for ffplay is the original way where each media type has it's own UDP connection and the RTCP
// also require a separate UDP connection on RTP port + 1.
IPAddress localIPAddress = IPAddress.Any;
IPEndPoint audioRtpEP = dstBaseEndPoint;
IPEndPoint audioRtcpEP = new IPEndPoint(dstBaseEndPoint.Address, dstBaseEndPoint.Port + 1);
IPEndPoint videoRtpEP = new IPEndPoint(dstBaseEndPoint.Address, dstBaseEndPoint.Port + 2);
IPEndPoint videoRtcpEP = new IPEndPoint(dstBaseEndPoint.Address, dstBaseEndPoint.Port + 3);
RTPSession audioRtpSession = new RTPSession((int)RTPPayloadTypesEnum.PCMU, null, null);
RTPSession videoRtpSession = new RTPSession(VP8_PAYLOAD_TYPE_ID, null, null);
DateTime lastRtcpSenderReportSentAt = DateTime.Now;
NetServices.CreateRtpSocket(localIPAddress, RAW_RTP_START_PORT_RANGE, RAW_RTP_END_PORT_RANGE, true, out audioSrcRtpSocket, out audioSrcControlSocket);
NetServices.CreateRtpSocket(localIPAddress, ((IPEndPoint)audioSrcRtpSocket.LocalEndPoint).Port, RAW_RTP_END_PORT_RANGE, true, out videoSrcRtpSocket, out videoSrcControlSocket);
OnMediaSampleReady += (mediaType, timestamp, sample) =>
{
if (mediaType == MediaSampleTypeEnum.VP8)
{
videoRtpSession.SendVp8Frame(videoSrcRtpSocket, videoRtpEP, timestamp, sample);
}
else
{
audioRtpSession.SendAudioFrame(audioSrcRtpSocket, audioRtpEP, timestamp, sample);
}
// Deliver periodic RTCP sender reports. This helps the receiver to sync the audio and video stream timestamps.
// If there are gaps in the media, silence supression etc. then the sender repors shouldn't be triggered from the media samples.
// In this case the samples are from an mp4 file which provides a constant uninterrupted stream.
if (DateTime.Now.Subtract(lastRtcpSenderReportSentAt).TotalSeconds >= RTCP_SR_PERIOD_SECONDS)
{
videoRtpSession.SendRtcpSenderReport(videoSrcControlSocket, videoRtcpEP, _vp8Timestamp);
audioRtpSession.SendRtcpSenderReport(audioSrcControlSocket, audioRtcpEP, _mulawTimestamp);
lastRtcpSenderReportSentAt = DateTime.Now;
}
};
}
catch (Exception excp)
{
logger.Error("Exception SendSamplesAsRtp. " + excp);
}
}
public void SendRtcpSenderReports(uint audioTimestamp, uint videoTimestamp)
{
var connectedIceCandidate = Peer.LocalIceCandidates.Where(y => y.RemoteRtpEndPoint != null).FirstOrDefault();
if (connectedIceCandidate != null)
{
var srcRtpEndPoint = connectedIceCandidate.LocalRtpSocket;
var dstRtpEndPoint = connectedIceCandidate.RemoteRtpEndPoint;
_audioRtpSession.SendRtcpSenderReport(srcRtpEndPoint, dstRtpEndPoint, audioTimestamp);
_videoRtpSession.SendRtcpSenderReport(srcRtpEndPoint, dstRtpEndPoint, videoTimestamp);
}
}