private static SDP GetSDP(IPEndPoint rtpSocket, string rtpFlowAttribute)
{
var sdp = new SDP()
{
SessionId = Crypto.GetRandomInt(5).ToString(),
Address = rtpSocket.Address.ToString(),
SessionName = "sipsorcery",
Timing = "0 0",
Connection = new SDPConnectionInformation(rtpSocket.Address.ToString()),
};
var audioAnnouncement = new SDPMediaAnnouncement()
{
Media = SDPMediaTypesEnum.audio,
MediaFormats = new List <SDPMediaFormat>()
{
new SDPMediaFormat((int)SDPMediaFormatsEnum.PCMU, "PCMU", 8000)
}
};
audioAnnouncement.Port = rtpSocket.Port;
audioAnnouncement.ExtraAttributes.Add($"a={rtpFlowAttribute}");
sdp.Media.Add(audioAnnouncement);
return(sdp);
}
public RTPMediaSession(SDPMediaTypesEnum mediaType, int formatTypeID, AddressFamily addrFamily)
: base(mediaType, formatTypeID, addrFamily, false, false)
{
// Construct the local SDP. There are a number of assumptions being made here:
// PCMU audio, RTP event support etc.
var mediaFormat = new SDPMediaFormat(formatTypeID);
var mediaAnnouncement = new SDPMediaAnnouncement
{
Media = mediaType,
MediaFormats = new List <SDPMediaFormat> {
mediaFormat
},
MediaStreamStatus = MediaStreamStatusEnum.SendRecv,
Port = base.RtpChannel.RTPPort
};
if (mediaType == SDPMediaTypesEnum.audio)
{
// RTP event support.
int clockRate = mediaFormat.GetClockRate();
SDPMediaFormat rtpEventFormat = new SDPMediaFormat(DTMF_EVENT_PAYLOAD_ID);
rtpEventFormat.SetFormatAttribute($"{TELEPHONE_EVENT_ATTRIBUTE}/{clockRate}");
rtpEventFormat.SetFormatParameterAttribute("0-16");
mediaAnnouncement.MediaFormats.Add(rtpEventFormat);
}
MediaAnnouncements.Add(mediaAnnouncement);
}
/// <summary>
/// 获取接收方端口号码
/// </summary>
/// <param name="sdpStr">SDP</param>
/// <returns></returns>
public int GetReceivePort(string sdpStr, SDPMediaTypesEnum mediaType)
{
string[] sdpLines = sdpStr.Split('\n');
foreach (var line in sdpLines)
{
if (line.Trim().StartsWith("m="))
{
Match mediaMatch = Regex.Match(line.Substring(2).Trim(), @"(?<type>\w+)\s+(?<port>\d+)\s+(?<transport>\S+)\s+(?<formats>.*)$");
if (mediaMatch.Success)
{
SDPMediaAnnouncement announcement = new SDPMediaAnnouncement
{
Media = SDPMediaTypes.GetSDPMediaType(mediaMatch.Result("${type}"))
};
Int32.TryParse(mediaMatch.Result("${port}"), out announcement.Port);
announcement.Transport = mediaMatch.Result("${transport}");
announcement.ParseMediaFormats(mediaMatch.Result("${formats}"));
if (announcement.Media != mediaType)
{
continue;
}
return(announcement.Port);
}
}
}
return(0);
}
/// <summary>
/// 设置媒体参数请求(实时)
/// </summary>
/// <param name="localIp">本地ip</param>
/// <param name="mediaPort">rtp/rtcp媒体端口(10000/10001)</param>
/// <returns></returns>
private string SetMediaAudio(string localIp, int port, string audioId)
{
SDPConnectionInformation sdpConn = new SDPConnectionInformation(localIp);
SDP sdp = new SDP();
sdp.Version = 0;
sdp.SessionId = "0";
sdp.Username = audioId;
sdp.SessionName = CommandType.Play.ToString();
sdp.Connection = sdpConn;
sdp.Timing = "0 0";
sdp.Address = localIp;
SDPMediaFormat psFormat = new SDPMediaFormat(SDPMediaFormatsEnum.PS);
psFormat.IsStandardAttribute = false;
SDPMediaAnnouncement media = new SDPMediaAnnouncement();
media.Media = SDPMediaTypesEnum.audio;
media.MediaFormats.Add(psFormat);
media.AddExtra("a=sendonly");
media.AddExtra("y=0100000002");
//media.AddExtra("f=v/////a/1/8/1");
media.AddFormatParameterAttribute(psFormat.FormatID, psFormat.Name);
media.Port = port;
sdp.Media.Add(media);
return(sdp.ToString());
}
private static SDP GetSDP(IPEndPoint rtpSocket, RTPPayloadTypesEnum audioPayloadType)
{
int samplingFrequency = RTPPayloadTypes.GetSamplingFrequency(audioPayloadType);
var sdp = new SDP()
{
SessionId = Crypto.GetRandomInt(5).ToString(),
Address = rtpSocket.Address.ToString(),
SessionName = "sipsorcery",
Timing = "0 0",
Connection = new SDPConnectionInformation(rtpSocket.Address.ToString()),
};
var audioAnnouncement = new SDPMediaAnnouncement()
{
Media = SDPMediaTypesEnum.audio,
MediaFormats = new List <SDPMediaFormat>()
{
new SDPMediaFormat((int)audioPayloadType, "PCMU", samplingFrequency)
}
};
audioAnnouncement.Port = rtpSocket.Port;
audioAnnouncement.ExtraAttributes.Add("a=sendrecv");
audioAnnouncement.ExtraAttributes.Add($"a=rtpmap:{DTMF_EVENT_PAYLOAD_ID} telephone-event/{samplingFrequency}");
audioAnnouncement.ExtraAttributes.Add($"a=fmtp:{DTMF_EVENT_PAYLOAD_ID} 0-15");
sdp.Media.Add(audioAnnouncement);
return(sdp);
}
private static SDP GetSDP(IPEndPoint rtpSocket)
{
var sdp = new SDP(rtpSocket.Address)
{
SessionId = Crypto.GetRandomInt(5).ToString(),
SessionName = "sipsorcery",
Timing = "0 0",
Connection = new SDPConnectionInformation(rtpSocket.Address),
};
var audioAnnouncement = new SDPMediaAnnouncement()
{
Media = SDPMediaTypesEnum.audio,
MediaFormats = new List <SDPMediaFormat>()
{
new SDPMediaFormat((int)SDPMediaFormatsEnum.PCMU, "PCMU", 8000),
new SDPMediaFormat((int)SDPMediaFormatsEnum.G722, "G722", 8000)
}
};
audioAnnouncement.Port = rtpSocket.Port;
audioAnnouncement.MediaStreamStatus = MediaStreamStatusEnum.SendRecv;
sdp.Media.Add(audioAnnouncement);
return(sdp);
}
/// <summary>
/// 设置媒体参数请求(实时)
/// </summary>
/// <param name="localIp">本地ip</param>
/// <param name="mediaPort">rtp/rtcp媒体端口(10000/10001)</param>
/// <returns></returns>
private string SetMediaReq(string localIp, int[] mediaPort)
{
SDPConnectionInformation sdpConn = new SDPConnectionInformation(localIp);
SDP sdp = new SDP();
sdp.Version = 0;
sdp.SessionId = "0";
sdp.Username = _msgCore.LocalSIPId;
sdp.SessionName = CommandType.Play.ToString();
sdp.Connection = sdpConn;
sdp.Timing = "0 0";
sdp.Address = localIp;
SDPMediaFormat psFormat = new SDPMediaFormat(SDPMediaFormatsEnum.PS);
psFormat.IsStandardAttribute = false;
SDPMediaFormat h264Format = new SDPMediaFormat(SDPMediaFormatsEnum.H264);
h264Format.IsStandardAttribute = false;
SDPMediaAnnouncement media = new SDPMediaAnnouncement();
media.Media = SDPMediaTypesEnum.video;
media.MediaFormats.Add(psFormat);
media.MediaFormats.Add(h264Format);
media.AddExtra("a=recvonly");
media.AddFormatParameterAttribute(psFormat.FormatID, psFormat.Name);
media.AddFormatParameterAttribute(h264Format.FormatID, h264Format.Name);
media.Port = mediaPort[0];
sdp.Media.Add(media);
return(sdp.ToString());
}
public void Start(string endpoint)
{
this.endpoint = endpoint;
var caller = "1003";
var password = passwords[0];
var port = FreePort.FindNextAvailableUDPPort(15090);
rtpChannel = new RTPChannel
{
DontTimeout = true,
RemoteEndPoint = new IPEndPoint(IPAddress.Parse(asterisk), port)
};
rtpChannel.SetFrameType(FrameTypesEnum.Audio);
rtpChannel.ReservePorts(15000, 15090);
rtpChannel.OnFrameReady += RtpChannel_OnFrameReady;
uac = new SIPClientUserAgent(transport, null, null, null, null);
var uri = SIPURI.ParseSIPURIRelaxed($"{ endpoint }@{ asterisk }");
var from = (new SIPFromHeader(caller, new SIPURI(caller, asterisk, null), null)).ToString();
var random = Crypto.GetRandomInt(5).ToString();
var sdp = new SDP
{
Version = 2,
Username = "******",
SessionId = random,
Address = localIPEndPoint.Address.ToString(),
SessionName = "redfox_" + random,
Timing = "0 0",
Connection = new SDPConnectionInformation(publicIPAddress.ToString())
};
var announcement = new SDPMediaAnnouncement
{
Media = SDPMediaTypesEnum.audio,
MediaFormats = new List <SDPMediaFormat>()
{
new SDPMediaFormat((int)SDPMediaFormatsEnum.PCMU, "PCMU", 8000)
},
Port = rtpChannel.RTPPort
};
sdp.Media.Add(announcement);
var descriptor = new SIPCallDescriptor(caller, password, uri.ToString(), from, null, null, null, null, SIPCallDirection.Out, SDP.SDP_MIME_CONTENTTYPE, sdp.ToString(), null);
uac.CallTrying += Uac_CallTrying;
uac.CallRinging += Uac_CallRinging;
uac.CallAnswered += Uac_CallAnswered;
uac.CallFailed += Uac_CallFailed;
uac.Call(descriptor);
}
/// <summary>
/// Gets an SDP packet that can be used by VoIP clients to negotiate an audio connection. The SDP will only
/// offer PCMU since that's all I've gotten around to handling.
/// </summary>
/// <param name="usePublicIP">If true and the public IP address is available from the STUN client then
/// the public IP address will be used in the SDP otherwise the host machine's default IPv4 address will
/// be used.</param>
/// <returns>An SDP packet that can be used by a VoIP client when initiating a call.</returns>
public SDP GetSDP(bool usePublicIP)
{
IPAddress rtpIPAddress = (usePublicIP && SoftphoneSTUNClient.PublicIPAddress != null) ? SoftphoneSTUNClient.PublicIPAddress : _defaultLocalAddress;
var sdp = new SDP()
{
SessionId = Crypto.GetRandomInt(5).ToString(),
Address = rtpIPAddress.ToString(),
SessionName = "sipsorcery",
Timing = "0 0",
Connection = new SDPConnectionInformation(rtpIPAddress.ToString()),
};
if (_rtpAudioChannel != null)
{
var audioAnnouncement = new SDPMediaAnnouncement()
{
Media = SDPMediaTypesEnum.audio,
MediaFormats = new List <SDPMediaFormat>()
{
new SDPMediaFormat((int)SDPMediaFormatsEnum.PCMU, "PCMU", 8000)
}
};
audioAnnouncement.Port = _rtpAudioChannel.RTPPort;
sdp.Media.Add(audioAnnouncement);
}
if (_rtpVideoChannel != null)
{
var videoAnnouncement = new SDPMediaAnnouncement()
{
Media = SDPMediaTypesEnum.video,
MediaFormats = new List <SDPMediaFormat>()
{
new SDPMediaFormat(96, "VP8", 90000)
}
};
videoAnnouncement.Port = _rtpVideoChannel.RTPPort;
sdp.Media.Add(videoAnnouncement);
}
return(sdp);
}
public void InvalidPortInRemoteOfferTest()
{
logger.LogDebug("--> " + System.Reflection.MethodBase.GetCurrentMethod().Name);
logger.BeginScope(System.Reflection.MethodBase.GetCurrentMethod().Name);
var remoteOffer = new SDP();
var sessionPort = 5523;
var sessionEndpoint = "10vMB2Ee;tcp";
remoteOffer.Connection = new SDPConnectionInformation(IPAddress.Loopback);
var messageMediaFormat = new SDPMessageMediaFormat();
messageMediaFormat.IP = remoteOffer.Connection.ConnectionAddress;
messageMediaFormat.Port = sessionPort.ToString();
messageMediaFormat.Endpoint = sessionEndpoint;
messageMediaFormat.AcceptTypes = new List <string>
{
"text/plain",
"text/x-msrp-heartbeat"
};
SDPMediaAnnouncement messageAnnouncement = new SDPMediaAnnouncement(
SDPMediaTypesEnum.message,
remoteOffer.Connection,
sessionPort,
messageMediaFormat);
messageAnnouncement.Transport = "TCP/MSRP";
remoteOffer.Media.Add(messageAnnouncement);
var sdpOffer = remoteOffer.ToString();
var msrpMediaAttribute =
$"{SDPMediaAnnouncement.MEDIA_FORMAT_PATH_MSRP_PREFIX}//{remoteOffer.Connection.ConnectionAddress}:{sessionPort}/{sessionEndpoint}";
var msrpMediaTypes = $"{SDPMediaAnnouncement.MEDIA_FORMAT_PATH_ACCEPT_TYPES_PREFIX}text/plain text/x-msrp-heartbeat";
var mediaDescription = $"m=message {sessionPort} TCP/MSRP *";
Assert.Contains(msrpMediaAttribute, sdpOffer);
Assert.Contains(msrpMediaTypes, sdpOffer);
Assert.Contains(mediaDescription, sdpOffer);
}
public Task <SDP> createOffer(RTCOfferOptions options)
{
SDP offerSdp = new SDP(IPAddress.Loopback);
offerSdp.SessionId = Crypto.GetRandomInt(5).ToString();
offerSdp.Connection = new SDPConnectionInformation(IPAddress.Loopback);
SDPMediaAnnouncement audioAnnouncement = new SDPMediaAnnouncement(
SDPMediaTypesEnum.audio,
1234,
new List <SDPMediaFormat> {
new SDPMediaFormat(SDPMediaFormatsEnum.PCMU)
});
audioAnnouncement.Transport = RTP_MEDIA_PROFILE;
offerSdp.Media.Add(audioAnnouncement);
return(Task.FromResult(offerSdp));
}
public SDP CreateAnswer(IPAddress connectionAddress)
{
SDP answerSdp = new SDP(IPAddress.Loopback);
answerSdp.SessionId = Crypto.GetRandomInt(5).ToString();
answerSdp.Connection = new SDPConnectionInformation(connectionAddress ?? IPAddress.Loopback);
SDPMediaAnnouncement audioAnnouncement = new SDPMediaAnnouncement(
SDPMediaTypesEnum.audio,
1234,
new List <SDPMediaFormat> {
new SDPMediaFormat(SDPMediaFormatsEnum.PCMU)
});
audioAnnouncement.Transport = RTP_MEDIA_PROFILE;
answerSdp.Media.Add(audioAnnouncement);
return(answerSdp);
}
public void InvalidPortInRemoteOfferTest()
{
logger.LogDebug("--> " + System.Reflection.MethodBase.GetCurrentMethod().Name);
logger.BeginScope(System.Reflection.MethodBase.GetCurrentMethod().Name);
RTPSession localSession = new RTPSession(false, false, false);
MediaStreamTrack localAudioTrack = new MediaStreamTrack(SDPMediaTypesEnum.audio, false, new List <SDPAudioVideoMediaFormat> {
new SDPAudioVideoMediaFormat(SDPWellKnownMediaFormatsEnum.PCMU)
});
localSession.addTrack(localAudioTrack);
var remoteOffer = new SDP();
remoteOffer.SessionId = Crypto.GetRandomInt(5).ToString();
remoteOffer.Connection = new SDPConnectionInformation(IPAddress.Loopback);
SDPMediaAnnouncement audioAnnouncement = new SDPMediaAnnouncement(
SDPMediaTypesEnum.audio,
66000,
new List <SDPAudioVideoMediaFormat> {
new SDPAudioVideoMediaFormat(SDPWellKnownMediaFormatsEnum.PCMU)
});
audioAnnouncement.Transport = RTPSession.RTP_MEDIA_PROFILE;
remoteOffer.Media.Add(audioAnnouncement);
var result = localSession.SetRemoteDescription(SIP.App.SdpType.offer, remoteOffer);
logger.LogDebug($"Set remote description on local session result {result}.");
Assert.Null(localSession.AudioDestinationEndPoint);
localSession.Close("normal");
}
/// <summary>
/// 设置媒体参数请求(实时)
/// </summary>
/// <param name="localIp">本地ip</param>
/// <param name="mediaPort">rtp/rtcp媒体端口(10000/10001)</param>
/// <returns></returns>
private string SetMediaAudio(string localIp, int port, string audioId)
{
var sdpConn = new SDPConnectionInformation(localIp);
var sdp = new SDP
{
Version = 0,
SessionId = "0",
Username = audioId,
SessionName = CommandType.Play.ToString(),
Connection = sdpConn,
Timing = "0 0",
Address = localIp
};
var psFormat = new SDPMediaFormat(SDPMediaFormatsEnum.PS)
{
IsStandardAttribute = false
};
var media = new SDPMediaAnnouncement
{
Media = SDPMediaTypesEnum.audio
};
media.MediaFormats.Add(psFormat);
media.AddExtra("a=sendonly");
media.AddExtra("y=0100000002");
//media.AddExtra("f=v/////a/1/8/1");
media.AddFormatParameterAttribute(psFormat.FormatID, psFormat.Name);
media.Port = port;
sdp.Media.Add(media);
return(sdp.ToString());
}
private void Transport_SIPTransportRequestReceived(SIPEndPoint localSIPEndPoint, SIPEndPoint remoteEndPoint, SIPRequest sipRequest)
{
var endpoint = new SIPEndPoint(SIPProtocolsEnum.udp, publicIPAddress, localSIPEndPoint.Port);
if (sipRequest.Method == SIPMethodsEnum.INVITE)
{
if (transaction != null)
{
return;
}
logger.DebugFormat("{0} Incoming call from {1}", prefix, sipRequest.Header.From.FromURI.User);
transaction = transport.CreateUASTransaction(sipRequest, remoteEndPoint, endpoint, null);
agent = new SIPServerUserAgent(
transport,
null,
sipRequest.Header.From.FromURI.User,
null,
SIPCallDirection.In,
null,
null,
null,
transaction);
agent.CallCancelled += Agent_CallCancelled;
agent.TransactionComplete += Agent_TransactionComplete;
agent.Progress(SIPResponseStatusCodesEnum.Trying, null, null, null, null);
agent.Progress(SIPResponseStatusCodesEnum.Ringing, null, null, null, null);
var answer = SDP.ParseSDPDescription(agent.CallRequest.Body);
var address = IPAddress.Parse(answer.Connection.ConnectionAddress);
var port = answer.Media.FirstOrDefault(m => m.Media == SDPMediaTypesEnum.audio).Port;
var random = Crypto.GetRandomInt(5).ToString();
var sdp = new SDP
{
Version = 2,
Username = "******",
SessionId = random,
Address = localIPEndPoint.Address.ToString(),
SessionName = "redfox_" + random,
Timing = "0 0",
Connection = new SDPConnectionInformation(publicIPAddress.ToString())
};
rtpChannel = new RTPChannel
{
DontTimeout = true,
RemoteEndPoint = new IPEndPoint(address, port)
};
rtpChannel.SetFrameType(FrameTypesEnum.Audio);
// TODO Fix hardcoded ports
rtpChannel.ReservePorts(15000, 15090);
rtpChannel.OnFrameReady += Channel_OnFrameReady;
rtpChannel.Start();
// Send some setup parameters to punch a hole in the firewall/router
rtpChannel.SendRTPRaw(new byte[] { 80, 95, 198, 88, 55, 96, 225, 141, 215, 205, 185, 242, 00 });
rtpChannel.OnControlDataReceived += (b) => { logger.Debug($"{prefix} Control Data Received; {b.Length} bytes"); };
rtpChannel.OnControlSocketDisconnected += () => { logger.Debug($"{prefix} Control Socket Disconnected"); };
var announcement = new SDPMediaAnnouncement
{
Media = SDPMediaTypesEnum.audio,
MediaFormats = new List <SDPMediaFormat>()
{
new SDPMediaFormat((int)SDPMediaFormatsEnum.PCMU, "PCMU", 8000)
},
Port = rtpChannel.RTPPort
};
sdp.Media.Add(announcement);
SetState(State.Listening, sipRequest.Header.From.FromURI.User);
agent.Progress(SIPResponseStatusCodesEnum.Accepted, null, null, null, null);
agent.Answer(SDP.SDP_MIME_CONTENTTYPE, sdp.ToString(), null, SIPDialogueTransferModesEnum.NotAllowed);
SetState(State.Busy, "");
return;
}
if (sipRequest.Method == SIPMethodsEnum.BYE)
{
if (State != State.Busy)
{
return;
}
logger.DebugFormat("{0} Hangup from {1}", prefix, sipRequest.Header.From.FromURI.User);
var noninvite = transport.CreateNonInviteTransaction(sipRequest, remoteEndPoint, endpoint, null);
var response = SIPTransport.GetResponse(sipRequest, SIPResponseStatusCodesEnum.Ok, null);
noninvite.SendFinalResponse(response);
SetState(State.Finished, Endpoint);
rtpChannel.OnFrameReady -= Channel_OnFrameReady;
rtpChannel.Close();
agent.TransactionComplete -= Agent_TransactionComplete;
agent.CallCancelled -= Agent_CallCancelled;
agent = null;
transaction = null;
SetState(State.Ready, Endpoint);
return;
}
if (sipRequest.Method == SIPMethodsEnum.ACK)
{
}
if (sipRequest.Method == SIPMethodsEnum.CANCEL)
{
}
}
