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.");
}
}
}
public WebRtcConnection(WebRtcSession webRtcSession)
{
CreatedAt = DateTime.Now;
WebRtcSession = webRtcSession;
}
