/// <summary>
/// Initialises the RTP session state and starts the RTP channel UDP sockets.
/// </summary>
/// <param name="addrFamily">Whether the RTP channel should use IPv4 or IPv6.</param>
/// <param name="isRtcpMultiplexed">If true RTCP reports will be multiplexed with RTP on a single channel.
/// If false (standard mode) then a separate socket is used to send and receive RTCP reports.</param>
private void InitialiseRtpChannel(AddressFamily addrFamily, bool isRtcpMultiplexed)
{
var channelAddress = (addrFamily == AddressFamily.InterNetworkV6) ? IPAddress.IPv6Any : IPAddress.Any;
RtpChannel = new RTPChannel(channelAddress, !isRtcpMultiplexed);
RtpChannel.OnRTPDataReceived += RtpPacketReceived;
RtpChannel.OnClosed += OnRTPChannelClosed;
RtcpSession = new RTCPSession(m_sessionStreams.First().Ssrc, RtpChannel, isRtcpMultiplexed);
OnRtpPacketReceived += RtcpSession.RtpPacketReceived;
RtpChannel.OnControlDataReceived += RtcpSession.ControlDataReceived;
OnRtpPacketSent += RtcpSession.RtpPacketSent;
// Start the RTP and Control socket receivers and the RTCP session.
RtpChannel.Start();
RtcpSession.Start();
}
/// <summary>
/// Sets the Secure RTP (SRTP) delegates and marks this session as ready for communications.
/// </summary>
/// <param name="protectRtp">SRTP encrypt RTP packet delegate.</param>
/// <param name="unprotectRtp">SRTP decrypt RTP packet delegate.</param>
/// <param name="protectRtcp">SRTP encrypt RTCP packet delegate.</param>
/// <param name="unprotectRtcp">SRTP decrypt RTCP packet delegate.</param>
public void SetSecurityContext(
ProtectRtpPacket protectRtp,
ProtectRtpPacket unprotectRtp,
ProtectRtpPacket protectRtcp,
ProtectRtpPacket unprotectRtcp)
{
m_srtpProtect = protectRtp;
m_srtpUnprotect = unprotectRtp;
m_srtcpControlProtect = protectRtcp;
m_srtcpControlUnprotect = unprotectRtcp;
IsSecureContextReady = true;
// Start the reporting sessions.
m_audioRtcpSession?.Start();
m_videoRtcpSession?.Start();
logger.LogDebug("Secure context successfully set on RTPSession.");
}
/// <summary>
/// Initialises the RTP session state and starts the RTP channel UDP sockets.
/// </summary>
private void Initialise(AddressFamily addrFamily, ProtectRtpPacket srtcpProtect)
{
Ssrc = Convert.ToUInt32(Crypto.GetRandomInt(0, Int32.MaxValue));
SeqNum = Convert.ToUInt16(Crypto.GetRandomInt(0, UInt16.MaxValue));
RtpChannel = new RTPChannel((addrFamily == AddressFamily.InterNetworkV6) ? IPAddress.IPv6Any : IPAddress.Any, true);
MediaAnnouncement.Port = RtpChannel.RTPPort;
RtpChannel.OnRTPDataReceived += RtpPacketReceived;
RtpChannel.OnClosed += OnRTPChannelClosed;
RtcpSession = new RTCPSession(Ssrc, RtpChannel, srtcpProtect);
OnRtpPacketReceived += RtcpSession.RtpPacketReceived;
RtpChannel.OnControlDataReceived += RtcpSession.ControlDataReceived;
OnRtpPacketSent += RtcpSession.RtpPacketSent;
// Start the RTP and Control socket receivers and the RTCP session.
RtpChannel.Start();
RtcpSession.Start();
}
/// <summary>
/// Adds an additional RTP stream to this session. The effect of this is to multiplex
/// two or more RTP sessions on a single socket. Multiplexing is used by WebRTC.
/// </summary>
/// <param name="mediaType">The type of media for this stream. When multiplexing streams on an
/// RTP session there can be only one stream per media type.</param>
/// <param name="payloadTypeID">The payload type ID for this RTP stream. It's what gets set in the payload
/// type ID field in the RTP header.</param>
/// <param name="remotePayloadIDs">A list of the payload IDs the remote party can set in their RTP headers.</param>
public void AddStream(SDPMediaTypesEnum mediaType, int payloadTypeID, List <int> remotePayloadIDs)
{
if (!(mediaType == SDPMediaTypesEnum.audio || mediaType == SDPMediaTypesEnum.video))
{
throw new ApplicationException($"The RTPSession does not know how to transmit media type {mediaType}.");
}
else if (mediaType == SDPMediaTypesEnum.audio && m_audioStream != null)
{
m_audioStream.UpdateRemotePayloadIDs(remotePayloadIDs);
}
else if (mediaType == SDPMediaTypesEnum.video && m_videoStream != null)
{
m_videoStream.UpdateRemotePayloadIDs(remotePayloadIDs);
}
else
{
if (mediaType == SDPMediaTypesEnum.audio)
{
m_audioStream = new RTPSessionStream(SDPMediaTypesEnum.audio, payloadTypeID, remotePayloadIDs);
m_audioRtcpSession = new RTCPSession(SDPMediaTypesEnum.audio, m_audioStream.Ssrc);
m_audioRtcpSession.OnReportReadyToSend += SendRtcpReport;
if (!IsSecure)
{
m_audioRtcpSession.Start();
}
}
else if (mediaType == SDPMediaTypesEnum.video)
{
m_videoStream = new RTPSessionStream(SDPMediaTypesEnum.video, payloadTypeID, remotePayloadIDs);
m_videoRtcpSession = new RTCPSession(SDPMediaTypesEnum.video, m_videoStream.Ssrc);
m_videoRtcpSession.OnReportReadyToSend += SendRtcpReport;
if (!IsSecure)
{
m_videoRtcpSession.Start();
}
}
}
}