/// <summary>
/// Example of how to create a basic RTP session object and hook up the event handlers.
/// </summary>
/// <param name="ua">The suer agent the RTP session is being created for.</param>
/// <returns>A new RTP session object.</returns>
private static RtpAudioSession CreateRtpSession(SIPUserAgent ua)
{
var rtpAudioSession = new RtpAudioSession(AddressFamily.InterNetwork);
// Add the required audio capabilities to the RTP session. These will
// automatically get used when creating SDP offers/answers.
var pcma = new SDPMediaFormat(SDPMediaFormatsEnum.PCMA);
// RTP event support.
int clockRate = pcma.GetClockRate();
SDPMediaFormat rtpEventFormat = new SDPMediaFormat(DTMF_EVENT_PAYLOAD_ID);
rtpEventFormat.SetFormatAttribute($"{RTPSession.TELEPHONE_EVENT_ATTRIBUTE}/{clockRate}");
rtpEventFormat.SetFormatParameterAttribute("0-16");
var audioCapabilities = new List <SDPMediaFormat> {
pcma, rtpEventFormat
};
MediaStreamTrack audioTrack = new MediaStreamTrack(null, SDPMediaTypesEnum.audio, false, audioCapabilities);
rtpAudioSession.addTrack(audioTrack);
// Wire up the event handler for RTP packets received from the remote party.
rtpAudioSession.OnRtpPacketReceived += (type, rtp) => OnRtpPacketReceived(ua, type, rtp);
if (_sendSilenceTimer == null)
{
_sendSilenceTimer = new Timer(SendSilence, null, 0, SEND_SILENCE_PERIOD_MS);
}
return(rtpAudioSession);
}
public RTPMediaSession(SDPMediaTypesEnum mediaType, int formatTypeID, AddressFamily addrFamily)
: base(mediaType, formatTypeID, addrFamily, false, false)
{
// Construct the local SDP. There are a number of assumptions being made here:
// PCMU audio, RTP event support etc.
var mediaFormat = new SDPMediaFormat(formatTypeID);
var mediaAnnouncement = new SDPMediaAnnouncement
{
Media = mediaType,
MediaFormats = new List <SDPMediaFormat> {
mediaFormat
},
MediaStreamStatus = MediaStreamStatusEnum.SendRecv,
Port = base.RtpChannel.RTPPort
};
if (mediaType == SDPMediaTypesEnum.audio)
{
// RTP event support.
int clockRate = mediaFormat.GetClockRate();
SDPMediaFormat rtpEventFormat = new SDPMediaFormat(DTMF_EVENT_PAYLOAD_ID);
rtpEventFormat.SetFormatAttribute($"{TELEPHONE_EVENT_ATTRIBUTE}/{clockRate}");
rtpEventFormat.SetFormatParameterAttribute("0-16");
mediaAnnouncement.MediaFormats.Add(rtpEventFormat);
}
MediaAnnouncements.Add(mediaAnnouncement);
}
/// <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;
}
/// <summary>
/// Creates an audio only RTP session that can supply an audio stream to the caller.
/// </summary>
/// <param name="audioOptions">The options that determine the type of audio to stream to the remote party. Example
/// type of audio sources are music, silence, white noise etc.</param>
/// <param name="audioCodecs">The audio codecs to support.</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,d depending on system support, which minimises network routing
/// causing connection issues.</param>
public RtpAudioSession(AudioSourceOptions audioOptions, List <SDPMediaFormatsEnum> audioCodecs,
IPAddress bindAddress = null) :
base(false, false, false, bindAddress)
{
if (audioCodecs == null || audioCodecs.Count() == 0)
{
_audioCodecs = new List <SDPMediaFormatsEnum>
{
SDPMediaFormatsEnum.PCMU, SDPMediaFormatsEnum.PCMA, SDPMediaFormatsEnum.G722
};
}
else if (audioCodecs.Any(x =>
!(x == SDPMediaFormatsEnum.PCMU || x == SDPMediaFormatsEnum.PCMA || x == SDPMediaFormatsEnum.G722)))
{
throw new ApplicationException("Only PCMA, PCMU and G722 audio codecs are supported.");
}
_audioOpts = audioOptions;
_audioCodecs = audioCodecs ?? _audioCodecs;
var audioCapabilities = new List <SDPMediaFormat>();
foreach (var codec in _audioCodecs)
{
audioCapabilities.Add(new SDPMediaFormat(codec));
}
// RTP event support.
SDPMediaFormat rtpEventFormat = new SDPMediaFormat(DTMF_EVENT_PAYLOAD_ID);
rtpEventFormat.SetFormatAttribute($"{SDP.TELEPHONE_EVENT_ATTRIBUTE}/{RTP_TIMESTAMP_RATE}");
rtpEventFormat.SetFormatParameterAttribute("0-16");
audioCapabilities.Add(rtpEventFormat);
// Add a local audio track to the RTP session.
MediaStreamTrack audioTrack = new MediaStreamTrack(SDPMediaTypesEnum.audio, false, audioCapabilities);
base.addTrack(audioTrack);
}
/// <summary>
/// Creates a new basic RTP session that captures and renders audio to/from the default system devices.
/// </summary>
public NAudioRtpSession()
: base(false, false, false)
{
var pcmu = new SDPMediaFormat(SDPMediaFormatsEnum.PCMU);
var pcma = new SDPMediaFormat(SDPMediaFormatsEnum.PCMA);
// RTP event support.
int clockRate = pcmu.GetClockRate();
SDPMediaFormat rtpEventFormat = new SDPMediaFormat(DTMF_EVENT_PAYLOAD_ID);
rtpEventFormat.SetFormatAttribute($"{SDP.TELEPHONE_EVENT_ATTRIBUTE}/{clockRate}");
rtpEventFormat.SetFormatParameterAttribute("0-16");
var audioCapabilities = new List <SDPMediaFormat> {
pcmu, pcma, rtpEventFormat
};
MediaStreamTrack audioTrack = new MediaStreamTrack(null, SDPMediaTypesEnum.audio, false, audioCapabilities);
addTrack(audioTrack);
// Where the magic (for processing received media) happens.
base.OnRtpPacketReceived += RtpPacketReceived;
}
public RtpAudioSession(DummyAudioOptions audioOptions, List <SDPMediaFormatsEnum> audioCodecs) :
base(false, false, false)
{
if (audioCodecs == null || audioCodecs.Count() == 0)
{
_audioCodecs = new List <SDPMediaFormatsEnum> {
SDPMediaFormatsEnum.PCMA, SDPMediaFormatsEnum.PCMU, SDPMediaFormatsEnum.G722
};
}
else if (audioCodecs.Any(x => !(x == SDPMediaFormatsEnum.PCMU || x == SDPMediaFormatsEnum.PCMA || x == SDPMediaFormatsEnum.G722)))
{
throw new ApplicationException("Only PCMA, PCMU and G722 audio codecs are supported.");
}
_audioOpts = audioOptions;
_audioCodecs = audioCodecs;
// RTP event support.
SDPMediaFormat rtpEventFormat = new SDPMediaFormat(DTMF_EVENT_PAYLOAD_ID);
rtpEventFormat.SetFormatAttribute($"{SDP.TELEPHONE_EVENT_ATTRIBUTE}/{SAMPLE_RATE}");
rtpEventFormat.SetFormatParameterAttribute("0-16");
var audioCapabilities = new List <SDPMediaFormat> {
rtpEventFormat
};
foreach (var codec in _audioCodecs)
{
audioCapabilities.Add(new SDPMediaFormat(codec));
}
MediaStreamTrack audioTrack = new MediaStreamTrack(SDPMediaTypesEnum.audio, false, audioCapabilities);
base.addTrack(audioTrack);
}
/// <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,d depending on system support, which minimises network routing
/// causing connection issues.</param>
public RtpAVSession(AudioOptions audioOptions, VideoOptions videoOptions, IPAddress bindAddress = null)
: base(false, false, false, bindAddress)
{
_audioOpts = audioOptions ?? DefaultAudioOptions;
_videoOpts = videoOptions ?? DefaultVideoOptions;
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(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;
}