private void WebRtcAnswerReceived(WebSocketSharp.Net.WebSockets.WebSocketContext context, string webSocketID, string sdpAnswer)
{
try
{
logger.LogDebug("Answer SDP: " + sdpAnswer);
var answerSDP = SDP.ParseSDPDescription(sdpAnswer);
var conn = _webRtcConnections.Where(x => x.Key == webSocketID).Select(x => x.Value).SingleOrDefault();
if (conn.WebRtcSession == null)
{
logger.LogWarning("No WebRTC client entry exists for web socket ID " + webSocketID + ", ignoring.");
}
else
{
logger.LogDebug("New WebRTC client SDP answer for web socket ID " + webSocketID + ".");
conn.WebRtcSession.setRemoteDescription(SdpType.answer, answerSDP);
}
context.WebSocket.CloseAsync();
}
catch (Exception excp)
{
logger.LogError("Exception WebRtcAnswerReceived. " + excp.Message);
}
}
private void WebRtcStartCall(WebSocketSharp.Net.WebSockets.WebSocketContext context, string webSocketID)
{
logger.Debug("New WebRTC client added for web socket connection " + webSocketID + ".");
lock (_webRtcSessions)
{
if (!_webRtcSessions.Any(x => x.Key == webSocketID))
{
var webRtcSession = new WebRtcSession(webSocketID);
if (_webRtcSessions.TryAdd(webSocketID, webRtcSession))
{
webRtcSession.Peer.OnSdpOfferReady += (sdp) => { logger.Debug("Offer SDP: " + sdp); context.WebSocket.Send(sdp); };
webRtcSession.Peer.OnDtlsPacket += webRtcSession.DtlsPacketReceived;
webRtcSession.Peer.OnMediaPacket += webRtcSession.MediaPacketReceived;
webRtcSession.Peer.Initialise(DTLS_CERTIFICATE_THUMBRPINT, null);
webRtcSession.Peer.OnClose += () => { PeerClosed(webSocketID); };
}
else
{
logger.Error("Failed to add new WebRTC client to sessions dictionary.");
}
}
}
}
private void WebRtcStartCall(WebSocketSharp.Net.WebSockets.WebSocketContext context, string webSocketID, IPAddress defaultIPAddress, bool isEncryptionDisabled, MediaSourceEnum mediaSource)
{
logger.Debug($"New WebRTC client added for web socket connection {webSocketID} and local IP address {defaultIPAddress}, encryption disabled {isEncryptionDisabled}.");
var mediaTypes = new List <RtpMediaTypesEnum> {
RtpMediaTypesEnum.Video, RtpMediaTypesEnum.Audio
};
_mfSampleGrabber.Start(); // Does nothing if media session is not paused.
lock (_webRtcSessions)
{
if (!_webRtcSessions.Any(x => x.Key == webSocketID))
{
var webRtcSession = new WebRtcSession(_dtlsCertificatePath, _dtlsKeyPath, webSocketID, isEncryptionDisabled, mediaSource);
string dtlsThumbrpint = (isEncryptionDisabled == false) ? _dtlsCertificateThumbprint : null;
if (_webRtcSessions.TryAdd(webSocketID, webRtcSession))
{
webRtcSession.Peer.OnSdpOfferReady += (sdp) => { logger.Debug("Offer SDP: " + sdp); context.WebSocket.Send(sdp); };
webRtcSession.Peer.OnMediaPacket += webRtcSession.MediaPacketReceived;
webRtcSession.Peer.Initialise(dtlsThumbrpint, null, mediaTypes, defaultIPAddress, isEncryptionDisabled);
webRtcSession.Peer.OnClose += () => { PeerClosed(webSocketID); };
if (isEncryptionDisabled == false)
{
webRtcSession.Peer.OnDtlsPacket += webRtcSession.DtlsPacketReceived;
}
else
{
webRtcSession.Peer.OnIceConnected += webRtcSession.InitEncryptionDisabledSession;
}
}
else
{
logger.Error("Failed to add new WebRTC client.");
}
}
}
}
private async void WebRtcStartCall(WebSocketSharp.Net.WebSockets.WebSocketContext context, string webSocketID, MediaSourceEnum mediaSource)
{
logger.LogDebug($"New WebRTC client added for web socket connection {webSocketID}.");
if (!_webRtcConnections.Any(x => x.Key == webSocketID))
{
var webRtcSession = new WebRtcSession(AddressFamily.InterNetwork, _dtlsCertificateThumbprint, null, null);
webRtcSession.addTrack(SDPMediaTypesEnum.video, new List <SDPMediaFormat> {
new SDPMediaFormat(SDPMediaFormatsEnum.VP8)
});
// Don't need an audio track for the test pattern feed.
if (mediaSource == MediaSourceEnum.Max)
{
webRtcSession.addTrack(SDPMediaTypesEnum.audio, new List <SDPMediaFormat> {
new SDPMediaFormat(SDPMediaFormatsEnum.PCMU)
});
}
WebRtcConnection conn = new WebRtcConnection(webRtcSession);
if (_webRtcConnections.TryAdd(webSocketID, conn))
{
webRtcSession.OnClose += (reason) => PeerClosed(webSocketID, reason);
webRtcSession.RtpSession.OnRtcpBye += (reason) => PeerClosed(webSocketID, reason);
var offerSdp = await webRtcSession.createOffer();
webRtcSession.setLocalDescription(offerSdp);
logger.LogDebug($"Sending SDP offer to client {context.UserEndPoint}.");
context.WebSocket.Send(webRtcSession.SDP.ToString());
if (DoDtlsHandshake(webRtcSession))
{
if (mediaSource == MediaSourceEnum.Max)
{
OnMp4MediaSampleReady += conn.SendMedia;
if (!_isMp4Sampling)
{
_ = Task.Run(SampleMp4Media);
}
}
else if (mediaSource == MediaSourceEnum.TestPattern)
{
OnTestPatternSampleReady += conn.SendMedia;
if (!_isTestPatternSampling)
{
_ = Task.Run(SampleTestPattern);
}
}
}
else
{
PeerClosed(webSocketID, "dtls handshake failed");
}
}
else
{
logger.LogError("Failed to add new WebRTC client.");
}
}
}
