public WebRtcSession(string dtlsFingerprint, List <IPAddress> offerAddresses)
{
_dtlsCertificateFingerprint = dtlsFingerprint;
SessionID = Guid.NewGuid().ToString();
_rtpSession = new RTPSession((int)SDPMediaFormatsEnum.PCMU, AddressFamily.InterNetwork, true);
_videoSessionID = _rtpSession.AddStream(VP8_PAYLOAD_TYPE_ID, null);
_rtpChannel = _rtpSession.RtpChannel;
_rtpChannel.OnRTPDataReceived += OnRTPDataReceived;
_rtpSession.OnRtpClosed += Close;
_offerAddresses = offerAddresses;
}
/// <summary>
/// Default constructor.
/// </summary>
/// <param name="dtlsFingerprint">The fingerprint of our DTLS certificate (we always act as the DTLS server).
/// It gets placed in the SDP offer sent to the remote party.</param>
/// <param name="offerAddresses">Optional. A list of the IP addresses used as local ICE candidates.
/// If null then all local IP addresses get used.</param>
/// <param name="turnServerEndPoint">Optional. A parameter that can be used include a TURN
/// server in this session's ICE candidate gathering.</param>
public WebRtcSession(
AddressFamily addrFamily,
string dtlsFingerprint,
List <IPAddress> offerAddresses,
IPEndPoint turnServerEndPoint)
{
_dtlsCertificateFingerprint = dtlsFingerprint;
_offerAddresses = offerAddresses;
_turnServerEndPoint = turnServerEndPoint;
SessionID = Guid.NewGuid().ToString();
RtpSession = new RTPSession(addrFamily, true, true);
_rtpChannel = RtpSession.RtpChannel;
_rtpChannel.OnRTPDataReceived += OnRTPDataReceived;
RtpSession.OnRtpClosed += Close;
RtpSession.OnRtcpBye += Close;
}
/// <summary>
/// Default constructor.
/// </summary>
/// <param name="dtlsFingerprint">The fingerprint of our DTLS certificate (we always act as the DTLS server).
/// It gets placed in the SDP offer sent to the remote party.</param>
/// <param name="supportedAudioFormats">List of audio codecs that we support. Can be null or empty if
/// the session is not supporting audio.</param>
/// <param name="supportedVideoFormats">List of video codecs that we support. Can be null or empty if
/// the session is not supporting video.</param>
/// <param name="offerAddresses">Optional. A list of the IP addresses used as local ICE candidates.
/// If null then all local IP addresses get used.</param>
public WebRtcSession(
string dtlsFingerprint,
List <SDPMediaFormat> supportedAudioFormats,
List <SDPMediaFormat> supportedVideoFormats,
List <IPAddress> offerAddresses)
{
if (supportedAudioFormats == null && supportedVideoFormats == null)
{
throw new ApplicationException("At least one of the audio or video supported formats must be specified.");
}
_dtlsCertificateFingerprint = dtlsFingerprint;
_supportedAudioFormats = supportedAudioFormats;
_supportedVideoFormats = supportedVideoFormats;
SessionID = Guid.NewGuid().ToString();
if (_supportedAudioFormats != null && supportedAudioFormats.Count > 0)
{
RtpSession = new RTPSession(SDPMediaTypesEnum.audio, (int)supportedAudioFormats.First().FormatCodec, AddressFamily.InterNetwork, true, true);
}
else if (_supportedVideoFormats != null && supportedVideoFormats.Count > 0)
{
RtpSession = new RTPSession(SDPMediaTypesEnum.video, (int)supportedVideoFormats.First().FormatCodec, AddressFamily.InterNetwork, true, true);
}
if (RtpSession == null)
{
throw new ApplicationException("No supported audio or video types were provided.");
}
_rtpChannel = RtpSession.RtpChannel;
_rtpChannel.OnRTPDataReceived += OnRTPDataReceived;
RtpSession.OnRtpClosed += Close;
_offerAddresses = offerAddresses;
}
public void Start(string url)
{
_url = url;
Match urlMatch = Regex.Match(url, @"rtsp://(?<hostname>\S+?)/", RegexOptions.IgnoreCase);
if (!urlMatch.Success)
{
throw new ApplicationException("The URL provided to the RTSP client was not recognised, " + url + ".");
}
else
{
string hostname = urlMatch.Result("${hostname}");
int port = RTSP_PORT;
if (hostname.Contains(':'))
{
port = SIPSorcery.Sys.IPSocket.ParsePortFromSocket(hostname);
hostname = SIPSorcery.Sys.IPSocket.ParseHostFromSocket(hostname);
}
logger.LogDebug("RTSP client connecting to " + hostname + ", port " + port + ".");
_rtspConnection = new TcpClient(hostname, port);
_rtspStream = _rtspConnection.GetStream();
_rtpSession = new RTPSession();
_rtpSession.RTPPayloadHeaderLength = _rtpPayloadHeaderLength;
_rtpSession.ReservePorts();
_rtpSession.OnRTPQueueFull += RTPQueueFull;
RTSPRequest rtspRequest = new RTSPRequest(RTSPMethodsEnum.SETUP, url);
RTSPHeader rtspHeader = new RTSPHeader(_cseq++, null);
rtspHeader.Transport = new RTSPTransportHeader()
{
ClientRTPPortRange = _rtspSession.RTPPort + "-" + _rtspSession.ControlPort
};
rtspRequest.Header = rtspHeader;
string rtspReqStr = rtspRequest.ToString();
RTSPMessage rtspMessage = null;
System.Diagnostics.Debug.WriteLine(rtspReqStr);
byte[] rtspRequestBuffer = Encoding.UTF8.GetBytes(rtspReqStr);
_rtspStream.Write(rtspRequestBuffer, 0, rtspRequestBuffer.Length);
byte[] buffer = new byte[2048];
int bytesRead = _rtspStream.Read(buffer, 0, 2048);
if (bytesRead > 0)
{
System.Diagnostics.Debug.WriteLine(Encoding.UTF8.GetString(buffer, 0, bytesRead));
rtspMessage = RTSPMessage.ParseRTSPMessage(buffer, null, null);
if (rtspMessage.RTSPMessageType == RTSPMessageTypesEnum.Response)
{
var setupResponse = RTSPResponse.ParseRTSPResponse(rtspMessage);
if (setupResponse.Status == RTSPResponseStatusCodesEnum.OK)
{
_rtspSession.SessionID = setupResponse.Header.Session;
_rtspSession.RemoteEndPoint = new IPEndPoint((_rtspConnection.Client.RemoteEndPoint as IPEndPoint).Address, setupResponse.Header.Transport.GetServerRTPPort());
_rtspSession.Start();
logger.LogDebug("RTSP Response received to SETUP: " + setupResponse.Status + ", session ID " + _rtspSession.SessionID + ", server RTP endpoint " + _rtspSession.RemoteEndPoint + ".");
if (OnSetupSuccess != null)
{
OnSetupSuccess(this);
}
}
else
{
logger.LogWarning("RTSP Response received to SETUP: " + setupResponse.Status + ".");
throw new ApplicationException("An error response of " + setupResponse.Status + " was received for an RTSP setup request.");
}
}
}
else
{
throw new ApplicationException("Zero bytes were read from the RTSP client socket in response to a SETUP request.");
}
}
}
