/// <summary>
/// This class manages different media renderer's that can be included in a call, e.g. audio and video.
/// </summary>
/// <param name="dispatcher">Need a UI dispatcher so tasks can be executed on the UI thread. For example this object
/// gets created when a button is clicked on and is therefore owned by the UI thread. When a call transfer completes the
/// resources need to be closed without any UI interaction. In that case need to marshal back to the UI thread.</param>
/// <param name="useVideo">Set to true if the current call is going to be using video.</param>
public MediaManager(Dispatcher dispatcher, bool useVideo = false)
{
_dispatcher = dispatcher;
_useVideo = useVideo;
if (_useVideo)
{
_vpxDecoder = new VpxEncoder();
_vpxDecoder.InitDecoder();
_imageConverter = new ImageConvert();
}
}
/// <summary>
/// Creates a new RTP audio visual session with audio/video capturing and rendering capabilities.
/// </summary>
/// <param name="addrFamily">The address family to create the underlying socket on (IPv4 or IPv6).</param>
/// <param name="audioOptions">Options for the send and receive audio streams on this session.</param>
/// <param name="videoOptions">Options for the send and receive video streams on this session</param>
public RtpAVSession(AddressFamily addrFamily, AudioOptions audioOptions, VideoOptions videoOptions)
: base(addrFamily, false, false, false)
{
_audioOpts = audioOptions ?? DefaultAudioOptions;
_videoOpts = videoOptions ?? DefaultVideoOptions;
// Initialise the video decoding objects. Even if we are not sourcing video
// we need to be ready to receive and render.
_vpxDecoder = new VpxEncoder();
int res = _vpxDecoder.InitDecoder();
if (res != 0)
{
throw new ApplicationException("VPX decoder initialisation failed.");
}
_imgConverter = new ImageConvert();
if (_audioOpts.AudioSource != AudioSourcesEnum.None)
{
var pcmu = new SDPMediaFormat(SDPMediaFormatsEnum.PCMU);
// RTP event support.
int clockRate = pcmu.GetClockRate();
SDPMediaFormat rtpEventFormat = new SDPMediaFormat(DTMF_EVENT_PAYLOAD_ID);
rtpEventFormat.SetFormatAttribute($"{TELEPHONE_EVENT_ATTRIBUTE}/{clockRate}");
rtpEventFormat.SetFormatParameterAttribute("0-16");
var audioCapabilities = new List <SDPMediaFormat> {
pcmu, rtpEventFormat
};
MediaStreamTrack audioTrack = new MediaStreamTrack(null, SDPMediaTypesEnum.audio, false, audioCapabilities);
addTrack(audioTrack);
}
if (_videoOpts.VideoSource != VideoSourcesEnum.None)
{
MediaStreamTrack videoTrack = new MediaStreamTrack(null, SDPMediaTypesEnum.video, false, new List <SDPMediaFormat> {
new SDPMediaFormat(SDPMediaFormatsEnum.VP8)
});
addTrack(videoTrack);
}
// Where the magic (for processing received media) happens.
base.OnRtpPacketReceived += RtpPacketReceived;
}
static void Main(string[] args)
{
AddConsoleLogger();
_vpxEncoder = new VpxEncoder();
int res = _vpxEncoder.InitDecoder();
if (res != 0)
{
throw new ApplicationException("VPX decoder initialisation failed.");
}
_imgConverter = new ImageConvert();
// Start web socket.
Console.WriteLine("Starting web socket server...");
_webSocketServer = new WebSocketServer(IPAddress.Any, WEBSOCKET_PORT, true);
_webSocketServer.SslConfiguration.ServerCertificate = new System.Security.Cryptography.X509Certificates.X509Certificate2(LOCALHOST_CERTIFICATE_PATH);
_webSocketServer.SslConfiguration.CheckCertificateRevocation = false;
_webSocketServer.AddWebSocketService <SDPExchange>("/", (sdpExchanger) =>
{
sdpExchanger.WebSocketOpened += WebSocketOpened;
sdpExchanger.OnMessageReceived += WebSocketMessageReceived;
});
_webSocketServer.Start();
Console.WriteLine($"Waiting for browser web socket connection to {_webSocketServer.Address}:{_webSocketServer.Port}...");
// Open a Window to display the video feed from the WebRTC peer.
_form = new Form();
_form.AutoSize = true;
_form.BackgroundImageLayout = ImageLayout.Center;
_picBox = new PictureBox
{
Size = new Size(640, 480),
Location = new Point(0, 0),
Visible = true
};
_form.Controls.Add(_picBox);
Application.EnableVisualStyles();
Application.Run(_form);
}
/// <summary>
/// Creates a new RTP audio visual session with audio/video capturing and rendering capabilities.
/// </summary>
/// <param name="addrFamily">The address family to create the underlying socket on (IPv4 or IPv6).</param>
/// <param name="audioOptions">Options for the send and receive audio streams on this session.</param>
/// <param name="videoOptions">Options for the send and receive video streams on this session</param>
/// <param name="bindAddress">Optional. If specified this address will be used as the bind address for any RTP
/// and control sockets created. Generally this address does not need to be set. The default behaviour
/// is to bind to [::] or 0.0.0.0, depending on system support, which minimises network routing
/// causing connection issues.</param>
/// <param name="disableExternalAudioSource">If true then no attempt will be made to use an external audio
/// source, e.g. microphone.</param>
public RtpAVSession(AudioOptions audioOptions, VideoOptions videoOptions, IPAddress bindAddress = null, bool disableExternalAudioSource = false)
: base(false, false, false, bindAddress)
{
_audioOpts = audioOptions ?? DefaultAudioOptions;
_videoOpts = videoOptions ?? DefaultVideoOptions;
_disableExternalAudioSource = disableExternalAudioSource;
if (_audioOpts != null && _audioOpts.AudioCodecs != null &&
_audioOpts.AudioCodecs.Any(x => !(x == SDPMediaFormatsEnum.PCMU || x == SDPMediaFormatsEnum.PCMA || x == SDPMediaFormatsEnum.G722)))
{
throw new ApplicationException("Only PCMA, PCMU and G722 are supported for audio codec options.");
}
// Initialise the video decoding objects. Even if we are not sourcing video
// we need to be ready to receive and render.
_vpxDecoder = new VpxEncoder();
int res = _vpxDecoder.InitDecoder();
if (res != 0)
{
throw new ApplicationException("VPX decoder initialisation failed.");
}
_imgConverter = new ImageConvert();
if (_audioOpts.AudioSource != AudioSourcesEnum.None)
{
var pcmu = new SDPMediaFormat(SDPMediaFormatsEnum.PCMU);
//// RTP event support.
//int clockRate = pcmu.GetClockRate();
//SDPMediaFormat rtpEventFormat = new SDPMediaFormat(DTMF_EVENT_PAYLOAD_ID);
//rtpEventFormat.SetFormatAttribute($"{SDP.TELEPHONE_EVENT_ATTRIBUTE}/{clockRate}");
//rtpEventFormat.SetFormatParameterAttribute("0-16");
var audioCapabilities = new List <SDPMediaFormat>();
if (_audioOpts.AudioCodecs == null || _audioOpts.AudioCodecs.Count == 0)
{
audioCapabilities.Add(pcmu);
}
else
{
foreach (var codec in _audioOpts.AudioCodecs)
{
audioCapabilities.Add(new SDPMediaFormat(codec));
}
}
//audioCapabilities.Add(rtpEventFormat);
if (audioCapabilities.Any(x => x.FormatCodec == SDPMediaFormatsEnum.G722))
{
_g722Encode = new G722Codec();
_g722EncodeState = new G722CodecState(64000, G722Flags.None);
_g722Decode = new G722Codec();
_g722DecodeState = new G722CodecState(64000, G722Flags.None);
}
MediaStreamTrack audioTrack = new MediaStreamTrack(SDPMediaTypesEnum.audio, false, audioCapabilities);
addTrack(audioTrack);
}
if (_videoOpts.VideoSource != VideoSourcesEnum.None)
{
MediaStreamTrack videoTrack = new MediaStreamTrack(SDPMediaTypesEnum.video, false, new List <SDPMediaFormat> {
new SDPMediaFormat(SDPMediaFormatsEnum.VP8)
});
addTrack(videoTrack);
}
// Where the magic (for processing received media) happens.
base.OnRtpPacketReceived += RtpPacketReceived;
}
