protected override async Task ExecuteAsync(CancellationToken stoppingToken)
{
// Set up media sources.
_maxSource = new FFmpegFileSource(MP4_PATH, true, new AudioEncoder());
_maxSource.Initialise();
_testPatternSource = new VideoTestPatternSource();
_testPatternEncoder = new FFmpegVideoEndPoint();
_testPatternSource.OnVideoSourceRawSample += _testPatternEncoder.ExternalVideoSourceRawSample;
_musicSource = new AudioExtrasSource(new AudioEncoder(),
new AudioSourceOptions {
AudioSource = AudioSourcesEnum.Music
});
// The same sources are used for all connected peers (broadcast) so the codecs need
// to be restricted to a single well supported option.
_musicSource.RestrictCodecs(new List <AudioCodecsEnum> {
AudioCodecsEnum.PCMU
});
_maxSource.RestrictCodecs(new List <VideoCodecsEnum> {
VIDEO_CODEC
});
_testPatternEncoder.RestrictCodecs(new List <VideoCodecsEnum> {
VIDEO_CODEC
});
// Start web socket.
_logger.LogInformation("Starting web socket server...");
WebSocketServer webSocketServer = null;
if (!string.IsNullOrWhiteSpace(_certificatePath))
{
if (!File.Exists(_certificatePath))
{
throw new ApplicationException($"Certificate path could not be found {_certificatePath}.");
}
webSocketServer = new WebSocketServer(IPAddress.Any, WEBSOCKET_PORT, true);
webSocketServer.SslConfiguration.ServerCertificate = new X509Certificate2(_certificatePath);
webSocketServer.SslConfiguration.CheckCertificateRevocation = false;
webSocketServer.SslConfiguration.EnabledSslProtocols = System.Security.Authentication.SslProtocols.Tls12;
}
else
{
webSocketServer = new WebSocketServer(IPAddress.Any, WEBSOCKET_PORT);
}
webSocketServer.AddWebSocketService <WebRTCWebSocketPeer>("/", (peer) => peer.CreatePeerConnection = () => CreatePeerConnection(null));
webSocketServer.AddWebSocketService <WebRTCWebSocketPeer>($"/{MAX_URL}",
(peer) => peer.CreatePeerConnection = () => CreatePeerConnection(MAX_URL));
webSocketServer.Start();
_logger.LogInformation($"Waiting for web socket connections on {webSocketServer.Address}:{webSocketServer.Port}...");
await Task.Delay(Timeout.Infinite, stoppingToken);
}
/// <summary>
/// An asynchronous task that attempts to initiate a new call to a listening UAS.
/// </summary>
/// <param name="sipTransport">The transport object to use for the send.</param>
/// <param name="dst">The destination end point to send the request to.</param>
/// <returns>True if the expected response was received, false otherwise.</returns>
private static async Task <bool> InitiateCallTaskAsync(SIPTransport sipTransport, SIPURI dst)
{
//UdpClient hepClient = new UdpClient(0, AddressFamily.InterNetwork);
try
{
//sipTransport.SIPRequestOutTraceEvent += (localEP, remoteEP, req) =>
//{
// logger.LogDebug($"Request sent: {localEP}->{remoteEP}");
// logger.LogDebug(req.ToString());
// //var hepBuffer = HepPacket.GetBytes(localEP, remoteEP, DateTimeOffset.Now, 333, "myHep", req.ToString());
// //hepClient.SendAsync(hepBuffer, hepBuffer.Length, "192.168.11.49", 9060);
//};
//sipTransport.SIPResponseInTraceEvent += (localEP, remoteEP, resp) =>
//{
// logger.LogDebug($"Response received: {localEP}<-{remoteEP}");
// logger.LogDebug(resp.ToString());
// //var hepBuffer = HepPacket.GetBytes(remoteEP, localEP, DateTimeOffset.Now, 333, "myHep", resp.ToString());
// //hepClient.SendAsync(hepBuffer, hepBuffer.Length, "192.168.11.49", 9060);
//};
var ua = new SIPUserAgent(sipTransport, null);
ua.ClientCallTrying += (uac, resp) => logger.LogInformation($"{uac.CallDescriptor.To} Trying: {resp.StatusCode} {resp.ReasonPhrase}.");
ua.ClientCallRinging += (uac, resp) => logger.LogInformation($"{uac.CallDescriptor.To} Ringing: {resp.StatusCode} {resp.ReasonPhrase}.");
ua.ClientCallFailed += (uac, err, resp) => logger.LogWarning($"{uac.CallDescriptor.To} Failed: {err}");
ua.ClientCallAnswered += (uac, resp) => logger.LogInformation($"{uac.CallDescriptor.To} Answered: {resp.StatusCode} {resp.ReasonPhrase}.");
var audioOptions = new AudioSourceOptions {
AudioSource = AudioSourcesEnum.Silence
};
var audioExtrasSource = new AudioExtrasSource(new AudioEncoder(), audioOptions);
audioExtrasSource.RestrictCodecs(new List <AudioCodecsEnum> {
AudioCodecsEnum.PCMU
});
var voipMediaSession = new VoIPMediaSession(new MediaEndPoints {
AudioSource = audioExtrasSource
});
var result = await ua.Call(dst.ToString(), null, null, voipMediaSession);
ua.Hangup();
await Task.Delay(200);
return(result);
}
catch (Exception excp)
{
logger.LogError($"Exception InitiateCallTaskAsync. {excp.Message}");
return(false);
}
}
static async Task Main()
{
Console.WriteLine("SIPSorcery Convert Audio");
Log = AddConsoleLogger();
//WaveFormatConversionStream converter = new WaveFormatConversionStream(_format_s16le48k, )
_waveFile = new WaveFileWriter("output_s16le48k.mp3", _format_s16le48k);
var sipTransport = new SIPTransport();
var userAgent = new SIPUserAgent(sipTransport, null);
userAgent.OnCallHungup += (dialog) =>
{
Console.WriteLine("Call hungup.");
_waveFile?.Close();
};
//EnableTraceLogs(sipTransport);
var audioOptions = new AudioSourceOptions {
AudioSource = AudioSourcesEnum.Silence
};
var audioCodecs = new List <AudioCodecsEnum> {
AudioCodecsEnum.PCMU
};
AudioExtrasSource audioExtrasSource = new AudioExtrasSource(new AudioEncoder(), audioOptions);
audioExtrasSource.RestrictCodecs(audioCodecs);
var rtpSession = new VoIPMediaSession(new MediaEndPoints {
AudioSource = audioExtrasSource
});
rtpSession.OnRtpPacketReceived += RtpSession_OnRtpPacketReceived;
// Place the call and wait for the result.
bool callResult = await userAgent.Call(DESTINATION, null, null, rtpSession);
Console.WriteLine($"Call result {((callResult) ? "success" : "failure")}.");
Console.WriteLine("press any key to exit...");
Console.Read();
if (userAgent.IsCallActive)
{
Console.WriteLine("Hanging up.");
userAgent.Hangup();
}
// Clean up.
sipTransport.Shutdown();
}
private VoIPMediaSession CreateMockVoIPMediaEndPoint(List <AudioCodecsEnum> supportedCodecs = null)
{
var audioSource = new AudioExtrasSource();
audioSource.RestrictCodecs(supportedCodecs);
MediaEndPoints mockEndPoints = new MediaEndPoints
{
AudioSource = audioSource
};
return(new VoIPMediaSession(mockEndPoints));
}
/// <summary>
/// Example of how to create a basic RTP session object and hook up the event handlers.
/// </summary>
/// <param name="ua">The user agent the RTP session is being created for.</param>
/// <param name="dst">THe destination specified on an incoming call. Can be used to
/// set the audio source.</param>
/// <returns>A new RTP session object.</returns>
private static VoIPMediaSession CreateRtpSession(SIPUserAgent ua, string dst)
{
List <AudioCodecsEnum> codecs = new List <AudioCodecsEnum> {
AudioCodecsEnum.PCMU, AudioCodecsEnum.PCMA, AudioCodecsEnum.G722
};
var audioSource = AudioSourcesEnum.SineWave;
if (string.IsNullOrEmpty(dst) || !Enum.TryParse <AudioSourcesEnum>(dst, out audioSource))
{
audioSource = AudioSourcesEnum.Silence;
}
Log.LogInformation($"RTP audio session source set to {audioSource}.");
AudioExtrasSource audioExtrasSource = new AudioExtrasSource(new AudioEncoder(), new AudioSourceOptions {
AudioSource = audioSource
});
audioExtrasSource.RestrictCodecs(codecs);
var rtpAudioSession = new VoIPMediaSession(new MediaEndPoints {
AudioSource = audioExtrasSource
});
rtpAudioSession.AcceptRtpFromAny = true;
// Wire up the event handler for RTP packets received from the remote party.
rtpAudioSession.OnRtpPacketReceived += (ep, type, rtp) => OnRtpPacketReceived(ua, type, rtp);
rtpAudioSession.OnTimeout += (mediaType) =>
{
if (ua?.Dialogue != null)
{
Log.LogWarning($"RTP timeout on call with {ua.Dialogue.RemoteTarget}, hanging up.");
}
else
{
Log.LogWarning($"RTP timeout on incomplete call, closing RTP session.");
}
ua.Hangup();
};
return(rtpAudioSession);
}