/// <summary>
/// Creates a new RTP session. The synchronisation source and sequence number are initialised to
/// pseudo random values.
/// </summary>
/// <param name="formatTypeID">The format type ID for the media. It's what gets set in the payload
/// type ID field in the RTP header. A default media announcement will be created.</param>
/// <param name="srtpProtect">Optional secure DTLS context for encrypting RTP packets.</param>
/// <param name="srtcpProtect">Optional secure DTLS context for encrypting RTCP packets.</param>
/// <param name="rtpEventSupport">True if RTP event sending and receiving should be supported.</param>
/// <param name="addrFamily">Determines whether the RTP channel will use an IPv4 or IPv6 socket.</param>
public RTPSession(int formatTypeID, ProtectRtpPacket srtpProtect, ProtectRtpPacket srtcpProtect, bool rtpEventSupport, AddressFamily addrFamily)
{
MediaFormat = new SDPMediaFormat(formatTypeID);
MediaAnnouncement = new SDPMediaAnnouncement
{
Media = SDPMediaTypesEnum.audio,
MediaFormats = new List <SDPMediaFormat> {
MediaFormat
},
MediaStreamStatus = MediaStreamStatusEnum.SendRecv
};
if (rtpEventSupport)
{
int clockRate = MediaFormat.GetClockRate();
SDPMediaFormat rtpEventFormat = new SDPMediaFormat(DTMF_EVENT_PAYLOAD_ID);
rtpEventFormat.SetFormatAttribute($"{TELEPHONE_EVENT_ATTRIBUTE}/{clockRate}");
rtpEventFormat.SetFormatParameterAttribute("0-16");
MediaAnnouncement.MediaFormats.Add(rtpEventFormat);
}
m_rtpEventSupport = rtpEventSupport;
FormatTypeID = formatTypeID;
SrtpProtect = srtpProtect;
Initialise(addrFamily, srtcpProtect);
}
/// <summary>
/// Creates an answer to an SDP offer from a remote peer.
/// </summary>
/// <remarks>
/// As specified in https://www.w3.org/TR/webrtc/#dom-rtcpeerconnection-createanswer and
/// https://tools.ietf.org/html/rfc3264#section-6.1.
/// </remarks>
/// <param name="options">Optional. If supplied the options will be used to apply additional
/// controls over the generated answer SDP.</param>
public RTCSessionDescriptionInit createAnswer(RTCAnswerOptions options)
{
if (remoteDescription == null)
{
throw new ApplicationException("The remote SDP must be set before an SDP answer can be created.");
}
else
{
var audioCapabilities = (AudioLocalTrack != null && AudioRemoteTrack != null) ?
SDPMediaFormat.GetCompatibleFormats(AudioLocalTrack.Capabilities, AudioRemoteTrack.Capabilities) : null;
var videoCapabilities = (VideoLocalTrack != null && VideoRemoteTrack != null) ?
SDPMediaFormat.GetCompatibleFormats(VideoLocalTrack.Capabilities, VideoRemoteTrack.Capabilities) : null;
List <MediaStreamTrack> localTracks = GetLocalTracks();
var answerSdp = createBaseSdp(localTracks, audioCapabilities, videoCapabilities);
if (answerSdp.Media.Any(x => x.Media == SDPMediaTypesEnum.audio))
{
var audioAnnouncement = answerSdp.Media.Where(x => x.Media == SDPMediaTypesEnum.audio).Single();
audioAnnouncement.AddExtra($"{ICE_SETUP_ATTRIBUTE}{IceRole}");
}
if (answerSdp.Media.Any(x => x.Media == SDPMediaTypesEnum.video))
{
var videoAnnouncement = answerSdp.Media.Where(x => x.Media == SDPMediaTypesEnum.video).Single();
videoAnnouncement.AddExtra($"{ICE_SETUP_ATTRIBUTE}{IceRole}");
}
RTCSessionDescriptionInit initDescription = new RTCSessionDescriptionInit
{
type = RTCSdpType.answer,
sdp = answerSdp.ToString()
};
return(initDescription);
}
}
