/// <summary>
/// Forwards media from the SIP session to the WebRTC session.
/// </summary>
/// <param name="mediaType">The type of media.</param>
/// <param name="rtpPacket">The RTP packet received on the SIP session.</param>
private static void ForwardMedia(SDPMediaTypesEnum mediaType, RTPPacket rtpPacket)
{
if (_peerConnection != null && mediaType == SDPMediaTypesEnum.audio)
{
_peerConnection.SendAudio((uint)rtpPacket.Payload.Length, rtpPacket.Payload);
}
}
/// <summary>
/// Forwards media from the SIP session to the WebRTC session.
/// </summary>
/// <param name="remote">The remote endpoint the RTP packet was received from.</param>
/// <param name="mediaType">The type of media.</param>
/// <param name="rtpPacket">The RTP packet received on the SIP session.</param>
private static void ForwardAudioToPeerConnection(IPEndPoint remote, SDPMediaTypesEnum mediaType, RTPPacket rtpPacket)
{
if (_peerConnection != null && _peerConnection.connectionState == RTCPeerConnectionState.connected &&
mediaType == SDPMediaTypesEnum.audio)
{
_peerConnection.SendAudio((uint)rtpPacket.Payload.Length, rtpPacket.Payload);
}
}
private static void AudioSource_OnAudioSourceEncodedSample(uint durationRtpUnits, byte[] sample)
{
PeerConnection.SendAudio(durationRtpUnits, sample);
if (audioSink != null)
{
audioSink.GotAudioRtp(null, 0, 0, 0, 0, false, sample);
}
}
