/// <summary>
/// Initialises the WebRTC session by carrying out the ICE connectivity steps and when complete
/// handing the RTP socket off for the DTLS handshake. Once the handshake is complete the session
/// is ready for to exchange encrypted RTP and RTCP packets.
/// </summary>
/// <param name="turnServerEndPoint">An optional parameter that can be used include a TURN
/// server in this session's ICE candidate gathering.</param>
public async Task Initialise(DoDtlsHandshakeDelegate doDtlsHandshake, IPEndPoint turnServerEndPoint)
{
try
{
_doDtlsHandshake = doDtlsHandshake;
_turnServerEndPoint = turnServerEndPoint;
DateTime startGatheringTime = DateTime.Now;
IceConnectionState = IceConnectionStatesEnum.Gathering;
await GetIceCandidatesAsync();
logger.LogDebug($"ICE gathering completed for in {DateTime.Now.Subtract(startGatheringTime).TotalMilliseconds:#}ms, candidate count {LocalIceCandidates.Count}.");
IceConnectionState = IceConnectionStatesEnum.GatheringComplete;
if (LocalIceCandidates.Count == 0)
{
logger.LogWarning("No local socket candidates were found for WebRTC call closing.");
Close("No local ICE candidates available.");
}
else
{
string localIceCandidateString = null;
foreach (var iceCandidate in LocalIceCandidates)
{
localIceCandidateString += iceCandidate.ToString();
}
LocalIceUser = LocalIceUser ?? Crypto.GetRandomString(20);
LocalIcePassword = LocalIcePassword ?? Crypto.GetRandomString(20) + Crypto.GetRandomString(20);
//var localIceCandidate = GetIceCandidatesForMediaType(RtpMediaTypesEnum.None).First();
var offerHeader = String.Format(_sdpOfferTemplate, Crypto.GetRandomInt(10).ToString());
string dtlsAttribute = String.Format(_dtlsFingerprint, _dtlsCertificateFingerprint);
string rtpSecurityDescriptor = RTP_MEDIA_SECURE_DESCRIPTOR;
var audioOffer = String.Format(_sdpAudioPcmOfferTemplate,
_rtpChannel.RTPPort,
rtpSecurityDescriptor,
IPAddress.Loopback,
localIceCandidateString.TrimEnd(),
LocalIceUser,
LocalIcePassword,
dtlsAttribute);
var videoOffer = String.Format(_sdpVideoOfferTemplate,
rtpSecurityDescriptor,
IPAddress.Loopback,
LocalIceUser,
LocalIcePassword,
dtlsAttribute);
string offer = offerHeader + audioOffer + videoOffer;
//logger.LogDebug("WebRTC Offer SDP: " + offer);
SDP = offer;
OnSdpOfferReady?.Invoke(offer);
}
// We may have received some remote candidates from the remote part SDP so perform an immediate STUN check.
// If there are no remote candidates this call will end up being a NOP.
SendStunConnectivityChecks(null);
if (_doDtlsHandshake != null)
{
_ = Task.Run(() =>
{
int result = _doDtlsHandshake(this, _rtpChannel.m_rtpSocket, out _rtpSession.SrtpProtect, out _rtpSession.RtcpSession.SrtcpProtect);
IsDtlsNegotiationComplete = (result == 0);
});
}
}
catch (Exception excp)
{
logger.LogError("Exception WebRtcPeer.Initialise. " + excp);
Close(excp.Message);
}
}
/// <summary>
/// Initialises the WebRTC session by carrying out the ICE connectivity steps and when complete
/// handing the RTP socket off for the DTLS handshake. Once the handshake is complete the session
/// is ready for to exchange encrypted RTP and RTCP packets.
/// </summary>
/// <param name="turnServerEndPoint">An optional parameter that can be used include a TURN
/// server in this session's ICE candidate gathering.</param>
public async Task Initialise(DoDtlsHandshakeDelegate doDtlsHandshake, IPEndPoint turnServerEndPoint)
{
try
{
_doDtlsHandshake = doDtlsHandshake;
_turnServerEndPoint = turnServerEndPoint;
DateTime startGatheringTime = DateTime.Now;
IceConnectionState = IceConnectionStatesEnum.Gathering;
await GetIceCandidatesAsync();
logger.LogDebug($"ICE gathering completed for in {DateTime.Now.Subtract(startGatheringTime).TotalMilliseconds:#}ms, candidate count {LocalIceCandidates.Count}.");
IceConnectionState = IceConnectionStatesEnum.GatheringComplete;
if (LocalIceCandidates.Count == 0)
{
logger.LogWarning("No local socket candidates were found for WebRTC call closing.");
Close("No local ICE candidates available.");
}
else
{
bool includeAudioOffer = _supportedAudioFormats?.Count() > 0;
bool includeVideoOffer = _supportedVideoFormats?.Count() > 0;
bool haveIceCandidatesBeenAdded = false;
bool isMediaBundle = includeAudioOffer && includeVideoOffer; // Is this SDP offer bundling audio and video on the same RTP connection.
string localIceCandidateString = null;
foreach (var iceCandidate in LocalIceCandidates)
{
localIceCandidateString += iceCandidate.ToString();
}
LocalIceUser = LocalIceUser ?? Crypto.GetRandomString(20);
LocalIcePassword = LocalIcePassword ?? Crypto.GetRandomString(20) + Crypto.GetRandomString(20);
SDP offerSdp = new SDP(IPAddress.Loopback);
offerSdp.SessionId = Crypto.GetRandomInt(5).ToString();
// Add a bundle attribute. Indicates that audio and video sessions will be multiplexed
// on a single RTP socket.
if (isMediaBundle)
{
offerSdp.Group = MEDIA_GROUPING;
}
if (includeAudioOffer)
{
SDPMediaAnnouncement audioAnnouncement = new SDPMediaAnnouncement(
SDPMediaTypesEnum.audio,
_rtpChannel.RTPPort,
_supportedAudioFormats);
audioAnnouncement.Transport = RTP_MEDIA_PROFILE;
if (!haveIceCandidatesBeenAdded)
{
audioAnnouncement.IceCandidates = LocalIceCandidates;
haveIceCandidatesBeenAdded = true;
}
audioAnnouncement.Connection = new SDPConnectionInformation(IPAddress.Any);
audioAnnouncement.IceUfrag = LocalIceUser;
audioAnnouncement.IcePwd = LocalIcePassword;
audioAnnouncement.DtlsFingerprint = _dtlsCertificateFingerprint;
audioAnnouncement.AddExtra(RTCP_MUX_ATTRIBUTE);
audioAnnouncement.AddExtra(SETUP_ATTRIBUTE);
audioAnnouncement.MediaStreamStatus = AudioStreamStatus;
if (isMediaBundle)
{
audioAnnouncement.MediaID = AUDIO_MEDIA_ID;
}
offerSdp.Media.Add(audioAnnouncement);
}
if (includeVideoOffer)
{
SDPMediaAnnouncement videoAnnouncement = new SDPMediaAnnouncement(
SDPMediaTypesEnum.video,
_rtpChannel.RTPPort,
_supportedVideoFormats);
videoAnnouncement.Transport = RTP_MEDIA_PROFILE;
if (!haveIceCandidatesBeenAdded)
{
videoAnnouncement.IceCandidates = LocalIceCandidates;
haveIceCandidatesBeenAdded = true;
}
videoAnnouncement.Connection = new SDPConnectionInformation(IPAddress.Any);
videoAnnouncement.IceUfrag = LocalIceUser;
videoAnnouncement.IcePwd = LocalIcePassword;
videoAnnouncement.DtlsFingerprint = _dtlsCertificateFingerprint;
videoAnnouncement.AddExtra(RTCP_MUX_ATTRIBUTE);
videoAnnouncement.AddExtra(SETUP_ATTRIBUTE);
videoAnnouncement.MediaStreamStatus = VideoStreamStatus;
if (isMediaBundle)
{
videoAnnouncement.MediaID = VIDEO_MEDIA_ID;
}
offerSdp.Media.Add(videoAnnouncement);
}
SDP = offerSdp;
OnSdpOfferReady?.Invoke(SDP);
}
// We may have received some remote candidates from the remote part SDP so perform an immediate STUN check.
// If there are no remote candidates this call will end up being a NOP.
SendStunConnectivityChecks(null);
if (_doDtlsHandshake != null)
{
_ = Task.Run(() =>
{
int result = _doDtlsHandshake(this);
IsDtlsNegotiationComplete = (result == 0);
});
}
}
catch (Exception excp)
{
logger.LogError("Exception WebRtcPeer.Initialise. " + excp);
Close(excp.Message);
}
}
