public async Task IncomingCallNoSdpUnitTest()
{
logger.LogDebug("--> " + System.Reflection.MethodBase.GetCurrentMethod().Name);
logger.BeginScope(System.Reflection.MethodBase.GetCurrentMethod().Name);
SIPTransport transport = new SIPTransport();
transport.AddSIPChannel(new MockSIPChannel(new System.Net.IPEndPoint(IPAddress.Any, 0)));
SIPUserAgent userAgent = new SIPUserAgent(transport, null);
string inviteReqStr = "INVITE sip:192.168.11.50:5060 SIP/2.0" + m_CRLF +
"Via: SIP/2.0/UDP 192.168.11.50:60163;rport;branch=z9hG4bKPj869f70960bdd4204b1352eaf242a3691" + m_CRLF +
"To: <sip:[email protected]>;tag=ZUJSXRRGXQ" + m_CRLF +
"From: <sip:[email protected]>;tag=4a60ce364b774258873ff199e5e39938" + m_CRLF +
"Call-ID: 17324d6df8744d978008c8997bfd208d" + m_CRLF +
"CSeq: 3532 INVITE" + m_CRLF +
"Contact: <sip:[email protected]:60163;ob>" + m_CRLF +
"Max-Forwards: 70" + m_CRLF +
"Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS" + m_CRLF +
"Supported: replaces, 100rel, timer, norefersub" + m_CRLF +
"Content-Length: 0" + m_CRLF +
"Content-Type: application/sdp" + m_CRLF +
"Session-Expires: 1800" + m_CRLF + m_CRLF;
SIPEndPoint dummySipEndPoint = new SIPEndPoint(new IPEndPoint(IPAddress.Loopback, 0));
SIPMessageBuffer sipMessageBuffer = SIPMessageBuffer.ParseSIPMessage(inviteReqStr, dummySipEndPoint, dummySipEndPoint);
SIPRequest inviteReq = SIPRequest.ParseSIPRequest(sipMessageBuffer);
var uas = userAgent.AcceptCall(inviteReq);
await userAgent.Answer(uas, CreateMockVoIPMediaEndPoint());
// The call attempt should timeout while waiting for the ACK request with the SDP answer.
Assert.False(userAgent.IsCallActive);
}
static void Main()
{
Console.WriteLine("SIPSorcery Getting Started Demo");
Log = AddConsoleLogger();
_waveFile = new WaveFileWriter("output.mp3", _waveFormat);
_sipTransport = new SIPTransport();
_sipTransport.AddSIPChannel(new SIPUDPChannel(new IPEndPoint(IPAddress.Any, SIP_LISTEN_PORT)));
var userAgent = new SIPUserAgent(_sipTransport, null, true);
userAgent.ServerCallCancelled += (uas) => Log.LogDebug("Incoming call cancelled by remote party.");
userAgent.OnCallHungup += (dialog) => _waveFile?.Close();
userAgent.OnIncomingCall += async(ua, req) =>
{
WindowsAudioEndPoint winAudioEP = new WindowsAudioEndPoint(new AudioEncoder());
VoIPMediaSession voipMediaSession = new VoIPMediaSession(winAudioEP.ToMediaEndPoints());
voipMediaSession.AcceptRtpFromAny = true;
voipMediaSession.OnRtpPacketReceived += OnRtpPacketReceived;
var uas = userAgent.AcceptCall(req);
await userAgent.Answer(uas, voipMediaSession);
};
Console.WriteLine("press any key to exit...");
Console.Read();
// Clean up.
_sipTransport.Shutdown();
}
/// <summary>
/// Because this is a server user agent the SIP transport must start listening for client user agents.
/// </summary>
private static async Task OnRequest(SIPEndPoint localSIPEndPoint, SIPEndPoint remoteEndPoint, SIPRequest sipRequest)
{
try
{
if (sipRequest.Header.From != null &&
sipRequest.Header.From.FromTag != null &&
sipRequest.Header.To != null &&
sipRequest.Header.To.ToTag != null)
{
// This is an in-dialog request that will be handled directly by a user agent instance.
}
else if (sipRequest.Method == SIPMethodsEnum.INVITE)
{
Log.LogInformation($"Incoming call request: {localSIPEndPoint}<-{remoteEndPoint} {sipRequest.URI}.");
SIPUserAgent ua = new SIPUserAgent(_sipTransport, null);
ua.OnCallHungup += OnHangup;
ua.ServerCallCancelled += (uas) => Log.LogDebug("Incoming call cancelled by remote party.");
ua.OnDtmfTone += (key, duration) => OnDtmfTone(ua, key, duration);
ua.OnRtpEvent += (evt, hdr) => Log.LogDebug($"rtp event {evt.EventID}, duration {evt.Duration}, end of event {evt.EndOfEvent}, timestamp {hdr.Timestamp}, marker {hdr.MarkerBit}.");
ua.OnTransactionTraceMessage += (tx, msg) => Log.LogDebug($"uas tx {tx.TransactionId}: {msg}");
ua.ServerCallRingTimeout += (uas) =>
{
Log.LogWarning($"Incoming call timed out in {uas.ClientTransaction.TransactionState} state waiting for client ACK, terminating.");
ua.Hangup();
};
var uas = ua.AcceptCall(sipRequest);
var rtpSession = CreateRtpSession(ua, sipRequest.URI.User);
await ua.Answer(uas, rtpSession);
if (ua.IsCallActive)
{
await rtpSession.Start();
_calls.TryAdd(ua.Dialogue.CallId, ua);
}
}
else if (sipRequest.Method == SIPMethodsEnum.BYE)
{
SIPResponse byeResponse = SIPResponse.GetResponse(sipRequest, SIPResponseStatusCodesEnum.CallLegTransactionDoesNotExist, null);
await _sipTransport.SendResponseAsync(byeResponse);
}
else if (sipRequest.Method == SIPMethodsEnum.SUBSCRIBE)
{
SIPResponse notAllowededResponse = SIPResponse.GetResponse(sipRequest, SIPResponseStatusCodesEnum.MethodNotAllowed, null);
await _sipTransport.SendResponseAsync(notAllowededResponse);
}
else if (sipRequest.Method == SIPMethodsEnum.OPTIONS || sipRequest.Method == SIPMethodsEnum.REGISTER)
{
SIPResponse optionsResponse = SIPResponse.GetResponse(sipRequest, SIPResponseStatusCodesEnum.Ok, null);
await _sipTransport.SendResponseAsync(optionsResponse);
}
}
catch (Exception reqExcp)
{
Log.LogWarning($"Exception handling {sipRequest.Method}. {reqExcp.Message}");
}
}
/// <summary>
/// Because this is a server user agent the SIP transport must start listening for client user agents.
/// </summary>
private static async Task OnRequest(SIPEndPoint localSIPEndPoint, SIPEndPoint remoteEndPoint, SIPRequest sipRequest)
{
try
{
if (sipRequest.Header.From != null &&
sipRequest.Header.From.FromTag != null &&
sipRequest.Header.To != null &&
sipRequest.Header.To.ToTag != null)
{
// This is an in-dialog request that will be handled directly by a user agent instance.
}
else if (sipRequest.Method == SIPMethodsEnum.INVITE)
{
Log.LogInformation($"Incoming call request: {localSIPEndPoint}<-{remoteEndPoint} {sipRequest.URI}.");
var userAgent = new SIPUserAgent(_sipTransport, null);
userAgent.ServerCallCancelled += (uas) => Log.LogDebug("Incoming call cancelled by remote party.");
userAgent.OnCallHungup += (dialog) => _waveFile?.Close();
var rtpSession = new RtpAVSession(
new AudioOptions
{
AudioSource = AudioSourcesEnum.CaptureDevice,
AudioCodecs = new List <SDPMediaFormatsEnum> {
SDPMediaFormatsEnum.PCMU, SDPMediaFormatsEnum.PCMA
}
},
null);
rtpSession.OnRtpPacketReceived += OnRtpPacketReceived;
var uas = userAgent.AcceptCall(sipRequest);
await userAgent.Answer(uas, rtpSession);
if (userAgent.IsCallActive)
{
await rtpSession.Start();
}
}
else if (sipRequest.Method == SIPMethodsEnum.BYE)
{
SIPResponse byeResponse = SIPResponse.GetResponse(sipRequest, SIPResponseStatusCodesEnum.CallLegTransactionDoesNotExist, null);
await _sipTransport.SendResponseAsync(byeResponse);
}
else if (sipRequest.Method == SIPMethodsEnum.SUBSCRIBE)
{
SIPResponse notAllowededResponse = SIPResponse.GetResponse(sipRequest, SIPResponseStatusCodesEnum.MethodNotAllowed, null);
await _sipTransport.SendResponseAsync(notAllowededResponse);
}
else if (sipRequest.Method == SIPMethodsEnum.OPTIONS || sipRequest.Method == SIPMethodsEnum.REGISTER)
{
SIPResponse optionsResponse = SIPResponse.GetResponse(sipRequest, SIPResponseStatusCodesEnum.Ok, null);
await _sipTransport.SendResponseAsync(optionsResponse);
}
}
catch (Exception reqExcp)
{
Log.LogWarning($"Exception handling {sipRequest.Method}. {reqExcp.Message}");
}
}
static void Main()
{
Console.WriteLine("SIPSorcery sip.js Demo");
Log = AddConsoleLogger();
var sipTransport = new SIPTransport();
EnableTraceLogs(sipTransport);
var sipChannel = new SIPWebSocketChannel(IPAddress.Loopback, 8081);
sipTransport.AddSIPChannel(sipChannel);
var userAgent = new SIPUserAgent(sipTransport, null, true);
userAgent.OnIncomingCall += async(ua, req) =>
{
Log.LogDebug($"Auto-answering incoming call from {req.Header.From}.");
var uas = userAgent.AcceptCall(req);
var peerConnection = new RTCPeerConnection(null);
peerConnection.onconnectionstatechange += (state) =>
{
Log.LogDebug($"Peer connection state change to {state}.");
if (state == RTCPeerConnectionState.failed)
{
peerConnection.Close("ice disconnection");
}
else if (state == RTCPeerConnectionState.connected)
{
peerConnection.OnRtpPacketReceived += OnRtpPacketReceived;
}
else if (state == RTCPeerConnectionState.closed)
{
peerConnection.OnRtpPacketReceived -= OnRtpPacketReceived;
}
};
MediaStreamTrack audioTrack = new MediaStreamTrack(new List <AudioFormat> {
new AudioFormat(SDPWellKnownMediaFormatsEnum.PCMU)
}, MediaStreamStatusEnum.SendRecv);
peerConnection.addTrack(audioTrack);
//MediaStreamTrack videoTrack = new MediaStreamTrack("1", SDPMediaTypesEnum.video, false, new List<SDPMediaFormat> { new SDPMediaFormat(SDPMediaFormatsEnum.VP8) }, MediaStreamStatusEnum.Inactive);
//peerConnection.addTrack(videoTrack);
var answerResult = await userAgent.Answer(uas, peerConnection);
};
Console.Write("press any key to exit...");
Console.Read();
sipTransport.Shutdown();
}
public async Task PlaceCallUnitTest()
{
logger.LogDebug("--> " + System.Reflection.MethodBase.GetCurrentMethod().Name);
logger.BeginScope(System.Reflection.MethodBase.GetCurrentMethod().Name);
SIPTransport serverTransport = new SIPTransport();
SIPUDPChannel udpChannel = new SIPUDPChannel(IPAddress.Loopback, 0);
serverTransport.AddSIPChannel(udpChannel);
// Set up two user agents: one to answer the test call and one to place it.
SIPUserAgent userAgentServer = new SIPUserAgent(serverTransport, null);
SIPUserAgent userAgentClient = new SIPUserAgent(new SIPTransport(), null);
serverTransport.SIPTransportRequestReceived += async(lep, rep, req) =>
{
logger.LogDebug("Request received: " + req.StatusLine);
var uas = userAgentServer.AcceptCall(req);
RtpAudioSession serverAudioSession = new RtpAudioSession(
new AudioSourceOptions {
AudioSource = AudioSourcesEnum.None
},
new List <SDPMediaFormatsEnum> {
SDPMediaFormatsEnum.PCMU
});
var answerResult = await userAgentServer.Answer(uas, serverAudioSession);
logger.LogDebug($"Server agent answer result {answerResult}.");
Assert.True(answerResult);
};
var dstUri = udpChannel.GetContactURI(SIPSchemesEnum.sip, new SIPEndPoint(SIPProtocolsEnum.udp, new IPEndPoint(IPAddress.Loopback, 0)));
logger.LogDebug($"Attempting call to {dstUri.ToString()}.");
RtpAudioSession clientAudioSession = new RtpAudioSession(
new AudioSourceOptions {
AudioSource = AudioSourcesEnum.None
},
new List <SDPMediaFormatsEnum> {
SDPMediaFormatsEnum.PCMU
});
var callResult = await userAgentClient.Call(dstUri.ToString(), null, null, clientAudioSession);
logger.LogDebug($"Client agent answer result {callResult }.");
Assert.True(callResult);
Assert.Equal(SIPDialogueStateEnum.Confirmed, userAgentClient.Dialogue.DialogueState);
Assert.Equal(SIPDialogueStateEnum.Confirmed, userAgentServer.Dialogue.DialogueState);
}
public async Task HandleInvalidSdpPortOnAnswerUnitTest()
{
logger.LogDebug("--> " + System.Reflection.MethodBase.GetCurrentMethod().Name);
logger.BeginScope(System.Reflection.MethodBase.GetCurrentMethod().Name);
SIPTransport transport = new SIPTransport();
SIPUserAgent userAgent = new SIPUserAgent(transport, null);
string inviteReqStr = @"INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 0.0.0.0;branch=z9hG4bK57441c4980b94e1686a06ae080be2935;rport
To: <sip:[email protected]>
From: <sip:0.0.0.0:0>;tag=MYILIYPHQD
Call-ID: ddf0e5a9687b4745925438da9000445d
CSeq: 1 INVITE
Max-Forwards: 70
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, PRACK, REFER, REGISTER, SUBSCRIBE
Content-Length: 0
v=0
o=- 1838015445 0 IN IP4 127.0.0.1
s=-
c=IN IP4 127.0.0.1
t=0 0
m=audio 79762 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=sendrecv";
SIPEndPoint dummySipEndPoint = new SIPEndPoint(new IPEndPoint(IPAddress.Any, 0));
SIPMessageBuffer sipMessageBuffer = SIPMessageBuffer.ParseSIPMessage(inviteReqStr, dummySipEndPoint, dummySipEndPoint);
SIPRequest inviteReq = SIPRequest.ParseSIPRequest(sipMessageBuffer);
var uas = userAgent.AcceptCall(inviteReq);
RTPSession rtpSession = new RTPSession(false, false, false);
MediaStreamTrack audioTrack = new MediaStreamTrack(SDPMediaTypesEnum.audio, false, new List <SDPMediaFormat> {
new SDPMediaFormat(SDPMediaFormatsEnum.PCMU)
});
rtpSession.addTrack(audioTrack);
var result = await userAgent.Answer(uas, rtpSession);
Assert.False(result);
rtpSession.Close("normal");
}
/// <summary>
/// Because this is a server user agent the SIP transport must start listening for client user agents.
/// </summary>
private static async Task OnRequest(SIPEndPoint localSIPEndPoint, SIPEndPoint remoteEndPoint, SIPRequest sipRequest)
{
try
{
if (sipRequest.Method == SIPMethodsEnum.INVITE)
{
Log.LogInformation($"Incoming call request: {localSIPEndPoint}<-{remoteEndPoint} {sipRequest.URI}.");
SIPUserAgent ua = new SIPUserAgent(_sipTransport, null);
ua.OnCallHungup += OnHangup;
ua.ServerCallCancelled += (uas) => Log.LogDebug("Incoming call cancelled by remote party.");
ua.OnDtmfTone += (key, duration) => OnDtmfTone(ua, key, duration);
var uas = ua.AcceptCall(sipRequest);
var rtpSession = CreateRtpSession(ua);
await ua.Answer(uas, rtpSession);
if (ua.IsCallActive)
{
_calls.TryAdd(ua.Dialogue.CallId, ua);
Timer sendSilenceTimer = new Timer(SendSilence, ua, 0, SEND_SILENCE_PERIOD_MS);
}
}
else if (sipRequest.Method == SIPMethodsEnum.BYE)
{
SIPResponse byeResponse = SIPResponse.GetResponse(sipRequest, SIPResponseStatusCodesEnum.CallLegTransactionDoesNotExist, null);
await _sipTransport.SendResponseAsync(byeResponse);
}
else if (sipRequest.Method == SIPMethodsEnum.SUBSCRIBE)
{
SIPResponse notAllowededResponse = SIPResponse.GetResponse(sipRequest, SIPResponseStatusCodesEnum.MethodNotAllowed, null);
await _sipTransport.SendResponseAsync(notAllowededResponse);
}
else if (sipRequest.Method == SIPMethodsEnum.OPTIONS || sipRequest.Method == SIPMethodsEnum.REGISTER)
{
SIPResponse optionsResponse = SIPResponse.GetResponse(sipRequest, SIPResponseStatusCodesEnum.Ok, null);
await _sipTransport.SendResponseAsync(optionsResponse);
}
}
catch (Exception reqExcp)
{
Log.LogWarning($"Exception handling {sipRequest.Method}. {reqExcp.Message}");
}
}
public async Task PlaceCallMismatchedCapabilitiesUnitTest()
{
logger.LogDebug("--> " + System.Reflection.MethodBase.GetCurrentMethod().Name);
logger.BeginScope(System.Reflection.MethodBase.GetCurrentMethod().Name);
SIPTransport serverTransport = new SIPTransport();
SIPUDPChannel udpChannel = new SIPUDPChannel(IPAddress.Loopback, 0);
serverTransport.AddSIPChannel(udpChannel);
// Set up two user agents: one to answer the test call and one to place it.
SIPUserAgent userAgentServer = new SIPUserAgent(serverTransport, null);
SIPUserAgent userAgentClient = new SIPUserAgent(new SIPTransport(), null);
serverTransport.SIPTransportRequestReceived += async(lep, rep, req) =>
{
logger.LogDebug("Request received: " + req.StatusLine);
var uas = userAgentServer.AcceptCall(req);
var serverAudioSession = CreateMockVoIPMediaEndPoint(new List <AudioCodecsEnum> {
AudioCodecsEnum.PCMU
});
var answerResult = await userAgentServer.Answer(uas, serverAudioSession);
logger.LogDebug($"Server agent answer result {answerResult}.");
Assert.False(answerResult);
};
var dstUri = udpChannel.GetContactURI(SIPSchemesEnum.sip, new SIPEndPoint(SIPProtocolsEnum.udp, new IPEndPoint(IPAddress.Loopback, 0)));
logger.LogDebug($"Attempting call to {dstUri.ToString()}.");
var clientMediaEndPoint = CreateMockVoIPMediaEndPoint(new List <AudioCodecsEnum> {
AudioCodecsEnum.G722
});
var callResult = await userAgentClient.Call(dstUri.ToString(), null, null, clientMediaEndPoint);
logger.LogDebug($"Client agent answer result {callResult }.");
Assert.False(callResult);
}
public async Task AnswerAudioOnlyUnitTest()
{
logger.LogDebug("--> " + System.Reflection.MethodBase.GetCurrentMethod().Name);
logger.BeginScope(System.Reflection.MethodBase.GetCurrentMethod().Name);
SIPTransport transport = new SIPTransport();
SIPUserAgent userAgent = new SIPUserAgent(transport, null);
string inviteReqStr = @"INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 0.0.0.0;branch=z9hG4bK57441c4980b94e1686a06ae080be2935;rport
To: <sip:[email protected]>
From: <sip:0.0.0.0:0>;tag=MYILIYPHQD
Call-ID: ddf0e5a9687b4745925438da9000445d
CSeq: 1 INVITE
Max-Forwards: 70
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, PRACK, REFER, REGISTER, SUBSCRIBE
Content-Length: 0
v=0
o=- 1838015445 0 IN IP4 127.0.0.1
s=-
c=IN IP4 127.0.0.1
t=0 0
m=audio 19762 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=sendrecv";
SIPEndPoint dummySipEndPoint = new SIPEndPoint(new IPEndPoint(IPAddress.Any, 0));
SIPMessageBuffer sipMessageBuffer = SIPMessageBuffer.ParseSIPMessage(inviteReqStr, dummySipEndPoint, dummySipEndPoint);
SIPRequest inviteReq = SIPRequest.ParseSIPRequest(sipMessageBuffer);
var uas = userAgent.AcceptCall(inviteReq);
var result = await userAgent.Answer(uas, CreateMediaSession());
Assert.True(result);
}
/// <summary>
/// Because this is a server user agent the SIP transport must start listening for client user agents.
/// </summary>
private static async Task OnRequest(SIPEndPoint localSIPEndPoint, SIPEndPoint remoteEndPoint, SIPRequest sipRequest)
{
try
{
if (sipRequest.Header.From != null &&
sipRequest.Header.From.FromTag != null &&
sipRequest.Header.To != null &&
sipRequest.Header.To.ToTag != null)
{
// This is an in-dialog request that will be handled directly by a user agent instance.
}
else if (sipRequest.Method == SIPMethodsEnum.INVITE)
{
Log.LogInformation($"Incoming call request: {localSIPEndPoint}<-{remoteEndPoint} {sipRequest.URI}.");
SIPUserAgent ua = new SIPUserAgent(_sipTransport, null);
ua.OnCallHungup += OnHangup;
ua.ServerCallCancelled += (uas) => Log.LogDebug("Incoming call cancelled by remote party.");
ua.OnDtmfTone += (key, duration) => OnDtmfTone(ua, key, duration);
ua.OnRtpEvent += (evt, hdr) => Log.LogDebug($"rtp event {evt.EventID}, duration {evt.Duration}, end of event {evt.EndOfEvent}, timestamp {hdr.Timestamp}, marker {hdr.MarkerBit}.");
ua.OnTransactionTraceMessage += (tx, msg) => Log.LogDebug($"uas tx {tx.TransactionId}: {msg}");
ua.ServerCallRingTimeout += (uas) =>
{
Log.LogWarning($"Incoming call timed out in {uas.ClientTransaction.TransactionState} state waiting for client ACK, terminating.");
ua.Hangup();
};
var uas = ua.AcceptCall(sipRequest);
var rtpSession = CreateRtpSession(ua);
// Insert a brief delay to allow testing of the "Ringing" progress response.
// Without the delay the call gets answered before it can be sent.
await Task.Delay(500);
await ua.Answer(uas, rtpSession);
if (ua.IsCallActive)
{
_calls.TryAdd(ua.Dialogue.CallId, ua);
if (sipRequest.URI.User != null)
{
if (Int32.TryParse(sipRequest.URI.User, out int dtmfCode))
{
Log.LogDebug($"URI dtmf code {dtmfCode}.");
while (dtmfCode > 0)
{
byte dtmfByte = (byte)(dtmfCode % 10);
Log.LogDebug($"Sending DTMF {dtmfByte} to caller.");
if (!ua.IsCallActive)
{
Log.LogWarning($"Client call no longer active.");
break;
}
else
{
await ua.SendDtmf(dtmfByte);
}
dtmfCode /= 10;
}
}
}
}
}
else if (sipRequest.Method == SIPMethodsEnum.BYE)
{
SIPResponse byeResponse = SIPResponse.GetResponse(sipRequest, SIPResponseStatusCodesEnum.CallLegTransactionDoesNotExist, null);
await _sipTransport.SendResponseAsync(byeResponse);
}
else if (sipRequest.Method == SIPMethodsEnum.SUBSCRIBE)
{
SIPResponse notAllowededResponse = SIPResponse.GetResponse(sipRequest, SIPResponseStatusCodesEnum.MethodNotAllowed, null);
await _sipTransport.SendResponseAsync(notAllowededResponse);
}
else if (sipRequest.Method == SIPMethodsEnum.OPTIONS || sipRequest.Method == SIPMethodsEnum.REGISTER)
{
SIPResponse optionsResponse = SIPResponse.GetResponse(sipRequest, SIPResponseStatusCodesEnum.Ok, null);
await _sipTransport.SendResponseAsync(optionsResponse);
}
}
catch (Exception reqExcp)
{
Log.LogWarning($"Exception handling {sipRequest.Method}. {reqExcp.Message}");
}
}
public async Task IncomingCallNoSdpWithACKUnitTest()
{
logger.LogDebug("--> " + System.Reflection.MethodBase.GetCurrentMethod().Name);
logger.BeginScope(System.Reflection.MethodBase.GetCurrentMethod().Name);
SIPTransport transport = new SIPTransport();
transport.AddSIPChannel(new MockSIPChannel(new System.Net.IPEndPoint(IPAddress.Any, 0)));
var dummySep = SIPEndPoint.ParseSIPEndPoint("udp:127.0.0.1:5060");
SIPUserAgent userAgent = new SIPUserAgent(transport, null);
string inviteReqStr = @"INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 127.0.0.1:51200;branch=z9hG4bKbeed9b0cde8d43cc8a2aae91526b6a1d;rport
To: <sip:[email protected]>
From: <sip:[email protected]>;tag=GCLNRILCDU
Call-ID: 7265e19f53a146a1bacdf4f4f8ea70b2
CSeq: 1 INVITE
Contact: <sip:127.0.0.1:51200>
Max-Forwards: 70
User-Agent: www.sipsorcery.com
Content-Length: 0
Content-Type: application/sdp" + m_CRLF + m_CRLF;
SIPEndPoint dummySipEndPoint = new SIPEndPoint(new IPEndPoint(IPAddress.Loopback, 0));
SIPMessageBuffer sipMessageBuffer = SIPMessageBuffer.ParseSIPMessage(inviteReqStr, dummySipEndPoint, dummySipEndPoint);
SIPRequest inviteReq = SIPRequest.ParseSIPRequest(sipMessageBuffer);
var uas = userAgent.AcceptCall(inviteReq);
var mediaSession = CreateMediaSession();
_ = Task.Run(() =>
{
Task.Delay(2000).Wait();
string ackReqStr = @"ACK sip:127.0.0.1:5060 SIP/2.0
Via: SIP/2.0/UDP 127.0.0.1:51200;branch=z9hG4bK76dfb1480ea14f778bd24afed1c8ded0;rport
To: <sip:[email protected]>;tag=YWPNZPMLPB
From: <sip:[email protected]>;tag=GCLNRILCDU
Call-ID: 7265e19f53a146a1bacdf4f4f8ea70b2
CSeq: 1 ACK
Max-Forwards: 70
Content-Length: 160
v=0
o=- 67424 0 IN IP4 127.0.0.1
s=-
c=IN IP4 127.0.0.1
t=0 0
m=audio 16976 RTP/AVP 8 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv" + m_CRLF + m_CRLF;
uas.ClientTransaction.ACKReceived(dummySep, dummySep, SIPRequest.ParseSIPRequest(ackReqStr));
});
await userAgent.Answer(uas, mediaSession);
Assert.True(userAgent.IsCallActive);
}
static void Main()
{
Console.WriteLine("SIPSorcery client user agent example.");
Console.WriteLine("Press ctrl-c to exit.");
CancellationTokenSource exitCts = new CancellationTokenSource(); // Cancellation token to stop the SIP transport and RTP stream.
AddConsoleLogger();
// Set up a default SIP transport.
var sipTransport = new SIPTransport();
sipTransport.AddSIPChannel(new SIPUDPChannel(new IPEndPoint(IPAddress.Any, SIP_LISTEN_PORT)));
// Un/comment this line to see/hide each SIP message sent and received.
EnableTraceLogs(sipTransport);
// To keep things a bit simpler this example only supports a single call at a time and the SIP server user agent
// acts as a singleton
SIPUserAgent userAgent = new SIPUserAgent(sipTransport, null);
CancellationTokenSource rtpCts = null; // Cancellation token to stop the RTP stream.
Socket rtpSocket = null;
Socket controlSocket = null;
// Because this is a server user agent the SIP transport must start listening for client user agents.
sipTransport.SIPTransportRequestReceived += (SIPEndPoint localSIPEndPoint, SIPEndPoint remoteEndPoint, SIPRequest sipRequest) =>
{
try
{
if (sipRequest.Header.From != null &&
sipRequest.Header.From.FromTag != null &&
sipRequest.Header.To != null &&
sipRequest.Header.To.ToTag != null)
{
userAgent.InDialogRequestReceivedAsync(sipRequest).Wait();
}
if (sipRequest.Method == SIPMethodsEnum.INVITE)
{
SIPSorcery.Sys.Log.Logger.LogInformation($"Incoming call request: {localSIPEndPoint}<-{remoteEndPoint} {sipRequest.URI}.");
// Check there's a codec we support in the INVITE offer.
var offerSdp = SDP.ParseSDPDescription(sipRequest.Body);
IPEndPoint dstRtpEndPoint = SDP.GetSDPRTPEndPoint(sipRequest.Body);
RTPSession rtpSession = null;
string audioFile = null;
if (offerSdp.Media.Any(x => x.Media == SDPMediaTypesEnum.audio && x.HasMediaFormat((int)RTPPayloadTypesEnum.PCMU)))
{
Log.LogDebug($"Using PCMU RTP media type and audio file {AUDIO_FILE_PCMU}.");
rtpSession = new RTPSession((int)RTPPayloadTypesEnum.PCMU, null, null);
audioFile = AUDIO_FILE_PCMU;
}
if (rtpSession == null)
{
// Didn't get a match on the codecs we support.
SIPResponse noMatchingCodecResponse = SIPTransport.GetResponse(sipRequest, SIPResponseStatusCodesEnum.NotAcceptableHere, null);
sipTransport.SendResponse(noMatchingCodecResponse);
}
else
{
// If there's already a call in progress hang it up. Of course this is not ideal for a real softphone or server but it
// means this example can be kept simpler.
if (userAgent?.IsAnswered == true)
{
userAgent?.Hangup();
}
rtpCts?.Cancel();
UASInviteTransaction uasTransaction = sipTransport.CreateUASTransaction(sipRequest, null);
if (userAgent.AcceptCall(uasTransaction))
{
rtpCts = new CancellationTokenSource();
// The RTP socket is listening on IPAddress.Any but the IP address placed into the SDP needs to be one the caller can reach.
IPAddress rtpAddress = NetServices.GetLocalAddressForRemote(dstRtpEndPoint.Address);
// Initialise an RTP session to receive the RTP packets from the remote SIP server.
NetServices.CreateRtpSocket(rtpAddress, RTP_PORT_START, RTP_PORT_END, false, out rtpSocket, out controlSocket);
var rtpRecvSession = new RTPSession((int)RTPPayloadTypesEnum.PCMU, null, null);
var rtpSendSession = new RTPSession((int)RTPPayloadTypesEnum.PCMU, null, null);
rtpSendSession.DestinationEndPoint = dstRtpEndPoint;
rtpRecvSession.OnReceiveFromEndPointChanged += (oldEP, newEP) =>
{
Log.LogDebug($"RTP destination end point changed from {oldEP} to {newEP}.");
rtpSendSession.DestinationEndPoint = newEP;
};
Task.Run(() => RecvRtp(rtpSocket, rtpRecvSession, rtpCts));
Task.Run(() => SendRtp(rtpSocket, rtpSendSession, rtpCts));
userAgent.Answer(GetSDP(rtpSocket.LocalEndPoint as IPEndPoint));
}
}
}
else if (sipRequest.Method == SIPMethodsEnum.SUBSCRIBE)
{
SIPResponse notAllowededResponse = SIPTransport.GetResponse(sipRequest, SIPResponseStatusCodesEnum.MethodNotAllowed, null);
sipTransport.SendResponse(notAllowededResponse);
}
else if (sipRequest.Method == SIPMethodsEnum.OPTIONS || sipRequest.Method == SIPMethodsEnum.REGISTER)
{
SIPResponse optionsResponse = SIPTransport.GetResponse(sipRequest, SIPResponseStatusCodesEnum.Ok, null);
sipTransport.SendResponse(optionsResponse);
}
}
catch (Exception reqExcp)
{
SIPSorcery.Sys.Log.Logger.LogWarning($"Exception handling {sipRequest.Method}. {reqExcp.Message}");
}
};
// Ctrl-c will gracefully exit the call at any point.
Console.CancelKeyPress += delegate(object sender, ConsoleCancelEventArgs e)
{
e.Cancel = true;
exitCts.Cancel();
rtpCts?.Cancel();
};
// At this point the call has been initiated and everything will be handled in an event handler.
Task.Run(async() =>
{
try
{
while (!exitCts.Token.WaitHandle.WaitOne(0))
{
var keyProps = Console.ReadKey();
if (keyProps.KeyChar == 't')
{
// Initiate a transfer.
bool transferResult = await userAgent.Transfer(SIPURI.ParseSIPURI(TRANSFER_DESTINATION_SIP_URI), new TimeSpan(0, 0, TRANSFER_TIMEOUT_SECONDS), exitCts.Token);
if (transferResult)
{
// If the transfer was accepted the original call will already have been hungup.
userAgent = null;
exitCts.Cancel();
}
else
{
Log.LogWarning($"Transfer to {TRANSFER_DESTINATION_SIP_URI} failed.");
}
}
else if (keyProps.KeyChar == 'q')
{
// Quit application.
exitCts.Cancel();
}
}
}
catch (Exception excp)
{
Log.LogError($"Exception Key Press listener. {excp.Message}.");
}
});
// Wait for a signal saying the call failed, was cancelled with ctrl-c or completed.
exitCts.Token.WaitHandle.WaitOne();
Log.LogInformation("Exiting...");
rtpSocket?.Close();
controlSocket?.Close();
if (userAgent != null)
{
if (userAgent.IsAnswered)
{
Log.LogInformation($"Hanging up call to {userAgent?.CallDescriptor?.To}.");
userAgent.Hangup();
}
// Give the final request time to be transmitted.
Log.LogInformation("Waiting 1s for call to clean up...");
Task.Delay(1000).Wait();
}
SIPSorcery.Net.DNSManager.Stop();
if (sipTransport != null)
{
Log.LogInformation("Shutting down SIP transport...");
sipTransport.Shutdown();
}
}
static async Task Main(string[] args)
{
Console.WriteLine("SIPSorcery Video Phone Command Line Demo");
Console.WriteLine("Press ctrl-c to exit.");
Log = AddConsoleLogger();
ManualResetEvent exitMRE = new ManualResetEvent(false);
ManualResetEvent waitForCallMre = new ManualResetEvent(false);
var parseResult = Parser.Default.ParseArguments <Options>(args);
_options = (parseResult as Parsed <Options>)?.Value;
if (parseResult.Tag != ParserResultType.NotParsed)
{
if (_options.ListCameras)
{
#region List webcams.
var webcams = await WindowsVideoEndPoint.GetVideoCatpureDevices();
if (webcams == null || webcams.Count == 0)
{
Console.WriteLine("No webcams were found.");
}
else
{
var index = 0;
foreach (var webcam in webcams)
{
Console.WriteLine($"{index}: \"{webcam.Name}\", use --cam={index}.");
index++;
}
}
#endregion
}
else if (_options.ListFormats != null)
{
#region List webcam formats.
var webcams = await WindowsVideoEndPoint.GetVideoCatpureDevices();
if (webcams == null || webcams.Count == 0)
{
Console.WriteLine("No webcams were found.");
}
else if (_options.ListFormats >= webcams.Count)
{
Console.WriteLine($"No webcam available for index {_options.ListFormats}.");
}
else
{
string webcamName = webcams[_options.ListFormats.Value].Name;
var formats = await WindowsVideoEndPoint.GetDeviceFrameFormats(webcamName);
Console.WriteLine($"Video frame formats for {webcamName}.");
foreach (var vidFmt in formats)
{
float vidFps = vidFmt.MediaFrameFormat.FrameRate.Numerator / vidFmt.MediaFrameFormat.FrameRate.Denominator;
string pixFmt = vidFmt.MediaFrameFormat.Subtype == WindowsVideoEndPoint.MF_I420_PIXEL_FORMAT ? "I420" : vidFmt.MediaFrameFormat.Subtype;
Console.WriteLine($"{vidFmt.Width}x{vidFmt.Height} {vidFps:0.##}fps {pixFmt}");
}
}
#endregion
}
else
{
string webcamName = null;
if (_options.WebcamIndex != null)
{
var webcams = await WindowsVideoEndPoint.GetVideoCatpureDevices();
if (webcams == null || webcams.Count == 0)
{
Console.WriteLine("No webcams were found.");
Application.Exit();
}
else if (webcams.Count < _options.WebcamIndex)
{
Console.WriteLine($"No webcam available for index {_options.WebcamIndex}.");
Application.Exit();
}
else
{
webcamName = webcams[_options.WebcamIndex.Value].Name;
Console.WriteLine($"Using webcam {webcamName}.");
}
}
_sipTransport = new SIPTransport();
if (string.IsNullOrEmpty(_options.CallDestination))
{
// We haven't been asked to place a call so we're listening.
IPAddress listenAddress = (System.Net.Sockets.Socket.OSSupportsIPv6) ? IPAddress.IPv6Any : IPAddress.Any;
var listenEndPoint = new IPEndPoint(listenAddress, SIP_PORT_DEFAULT);
try
{
SIPUDPChannel udpChannel = new SIPUDPChannel(listenEndPoint, true);
_sipTransport.AddSIPChannel(udpChannel);
}
catch (ApplicationException appExcp)
{
Console.WriteLine($"Failed to create UDP SIP channel on {listenEndPoint}, error {appExcp.Message}.");
SIPUDPChannel udpChannel = new SIPUDPChannel(new IPEndPoint(listenAddress, 0), true);
_sipTransport.AddSIPChannel(udpChannel);
}
var listeningEP = _sipTransport.GetSIPChannels().First().ListeningSIPEndPoint;
Console.WriteLine($"Listening for incoming call on {listeningEP}.");
}
EnableTraceLogs(_sipTransport);
// Open a window to display the video feed from the remote SIP party.
_form = new Form();
_form.Text = string.IsNullOrEmpty(_options.CallDestination) ? "Listener" : "Caller";
_form.AutoSize = true;
_form.BackgroundImageLayout = ImageLayout.Center;
_localVideoPicBox = new PictureBox
{
Size = new Size(VIDEO_FRAME_WIDTH, VIDEO_FRAME_HEIGHT),
Location = new Point(0, 0),
Visible = true
};
_remoteVideoPicBox = new PictureBox
{
Size = new Size(VIDEO_FRAME_WIDTH, VIDEO_FRAME_HEIGHT),
Location = new Point(0, VIDEO_FRAME_HEIGHT),
Visible = true
};
_form.Controls.Add(_localVideoPicBox);
_form.Controls.Add(_remoteVideoPicBox);
var userAgent = new SIPUserAgent(_sipTransport, null, true);
userAgent.OnCallHungup += (dialog) => exitMRE.Set();
WindowsAudioEndPoint windowsAudioEndPoint = null;
if (!_options.NoAudio)
{
windowsAudioEndPoint = new WindowsAudioEndPoint(new AudioEncoder());
windowsAudioEndPoint.RestrictFormats(x => x.Codec == AudioCodecsEnum.G722);
}
MediaEndPoints mediaEndPoints = null;
if (_options.TestPattern && _options.WebcamIndex == null)
{
var testPattern = new VideoTestPatternSource(new FFmpegVideoEncoder());
var decoderSink = new DecoderVideoSink(new FFmpegVideoEncoder());
//var decoderSink = new DecoderVideoSink(new VpxVideoEncoder());
testPattern.RestrictFormats(format => format.Codec == VIDEO_CODEC);
decoderSink.RestrictFormats(format => format.Codec == VIDEO_CODEC);
mediaEndPoints = new MediaEndPoints
{
AudioSink = windowsAudioEndPoint,
AudioSource = windowsAudioEndPoint,
VideoSink = decoderSink,
VideoSource = testPattern,
};
}
else
{
WindowsVideoEndPoint windowsVideoEndPoint = webcamName switch
{
null => new WindowsVideoEndPoint(new FFmpegVideoEncoder()),
_ => new WindowsVideoEndPoint(new FFmpegVideoEncoder(), webcamName),
};
windowsVideoEndPoint.RestrictFormats(format => format.Codec == VIDEO_CODEC);
mediaEndPoints = new MediaEndPoints
{
AudioSink = windowsAudioEndPoint,
AudioSource = windowsAudioEndPoint,
VideoSink = windowsVideoEndPoint,
VideoSource = windowsVideoEndPoint,
};
}
mediaEndPoints.VideoSource.OnVideoSourceRawSample += (uint durationMilliseconds, int width, int height, byte[] sample, VideoPixelFormatsEnum pixelFormat) =>
{
if (_isFormActivated)
{
_form?.BeginInvoke(new Action(() =>
{
if (_form.Handle != IntPtr.Zero)
{
int stride = width * 3;
if (pixelFormat == VideoPixelFormatsEnum.I420)
{
sample = PixelConverter.I420toBGR(sample, width, height, out stride);
}
if (_localVideoPicBox.Width != width || _localVideoPicBox.Height != height)
{
Log.LogDebug($"Adjusting local video display from {_localVideoPicBox.Width}x{_localVideoPicBox.Height} to {width}x{height}.");
_localVideoPicBox.Width = width;
_localVideoPicBox.Height = height;
}
unsafe
{
fixed(byte *s = sample)
{
System.Drawing.Bitmap bmpImage = new System.Drawing.Bitmap(width, height, stride, System.Drawing.Imaging.PixelFormat.Format24bppRgb, (IntPtr)s);
_localVideoPicBox.Image = bmpImage;
}
}
}
}));
}
};
Console.CancelKeyPress += delegate(object sender, ConsoleCancelEventArgs e)
{
e.Cancel = true;
Log.LogInformation("Exiting...");
waitForCallMre.Set();
exitMRE.Set();
};
if (string.IsNullOrEmpty(_options.CallDestination))
{
ActivateForm();
userAgent.OnIncomingCall += async(ua, req) =>
{
var voipMediaSession = new VoIPMediaSession(mediaEndPoints);
voipMediaSession.AcceptRtpFromAny = true;
if (voipMediaSession.VideoLocalTrack != null)
{
voipMediaSession.VideoLocalTrack.MaximumBandwidth = MAXIMUM_VIDEO_BANDWIDTH;
}
var uas = userAgent.AcceptCall(req);
await userAgent.Answer(uas, voipMediaSession);
Console.WriteLine("Starting local video source...");
await mediaEndPoints.VideoSource.StartVideo().ConfigureAwait(false);
waitForCallMre.Set();
};
Console.WriteLine("Waiting for incoming call...");
waitForCallMre.WaitOne();
}
else
{
var voipMediaSession = new VoIPMediaSession(mediaEndPoints);
voipMediaSession.AcceptRtpFromAny = true;
if (voipMediaSession.VideoLocalTrack != null)
{
voipMediaSession.VideoLocalTrack.MaximumBandwidth = MAXIMUM_VIDEO_BANDWIDTH;
}
ActivateForm();
Console.WriteLine("Starting local video source...");
await mediaEndPoints.VideoSource.StartVideo().ConfigureAwait(false);
// Place the call and wait for the result.
Task <bool> callTask = userAgent.Call(_options.CallDestination, null, null, voipMediaSession);
callTask.Wait(CALL_TIMEOUT_SECONDS * 1000);
}
if (userAgent.IsCallActive)
{
Log.LogInformation("Call attempt successful.");
mediaEndPoints.VideoSink.OnVideoSinkDecodedSample += (byte[] bmp, uint width, uint height, int stride, VideoPixelFormatsEnum pixelFormat) =>
{
if (_isFormActivated)
{
_form?.BeginInvoke(new Action(() =>
{
if (_form.Handle != IntPtr.Zero)
{
unsafe
{
if (_remoteVideoPicBox.Width != (int)width || _remoteVideoPicBox.Height != (int)height)
{
Log.LogDebug($"Adjusting remote video display from {_remoteVideoPicBox.Width}x{_remoteVideoPicBox.Height} to {width}x{height}.");
_remoteVideoPicBox.Width = (int)width;
_remoteVideoPicBox.Height = (int)height;
}
fixed(byte *s = bmp)
{
System.Drawing.Bitmap bmpImage = new System.Drawing.Bitmap((int)width, (int)height, stride, System.Drawing.Imaging.PixelFormat.Format24bppRgb, (IntPtr)s);
_remoteVideoPicBox.Image = bmpImage;
}
}
}
}));
}
};
}
else
{
Log.LogWarning("Call attempt failed.");
Console.WriteLine("Press ctrl-c to exit.");
}
exitMRE.WaitOne();
if (userAgent.IsCallActive)
{
Log.LogInformation("Hanging up.");
userAgent.Hangup();
}
Task.Delay(1000).Wait();
// Clean up.
if (_form.Handle != IntPtr.Zero)
{
_form.BeginInvoke(new Action(() => _form.Close()));
}
_sipTransport.Shutdown();
}
}
}
static void Main()
{
Console.WriteLine("SIPSorcery Call Hold and Blind Transfer example.");
Console.WriteLine("Press 'c' to initiate a call to the default destination.");
Console.WriteLine("Press 'h' to place an established call on and off hold.");
Console.WriteLine("Press 'H' to hangup an established call.");
Console.WriteLine("Press 't' to request a blind transfer on an established call.");
Console.WriteLine("Press 'q' or ctrl-c to exit.");
// Plumbing code to facilitate a graceful exit.
CancellationTokenSource exitCts = new CancellationTokenSource(); // Cancellation token to stop the SIP transport and RTP stream.
Log = AddConsoleLogger();
// Set up a default SIP transport.
var sipTransport = new SIPTransport();
sipTransport.AddSIPChannel(new SIPUDPChannel(new IPEndPoint(IPAddress.Any, SIP_LISTEN_PORT)));
Console.WriteLine($"Listening for incoming calls on: {sipTransport.GetSIPChannels().First().ListeningEndPoint}.");
EnableTraceLogs(sipTransport);
var winAudio = new WindowsAudioEndPoint(new AudioEncoder());
winAudio.RestrictCodecs(new List <AudioCodecsEnum> {
AudioCodecsEnum.PCMU
});
// Create a client/server user agent to place a call to a remote SIP server along with event handlers for the different stages of the call.
var userAgent = new SIPUserAgent(sipTransport, null, true);
userAgent.RemotePutOnHold += () => Log.LogInformation("Remote call party has placed us on hold.");
userAgent.RemoteTookOffHold += () => Log.LogInformation("Remote call party took us off hold.");
userAgent.OnIncomingCall += async(ua, req) =>
{
Log.LogInformation($"Incoming call from {req.Header.From.FriendlyDescription()} at {req.RemoteSIPEndPoint}.");
var uas = userAgent.AcceptCall(req);
if (userAgent?.IsCallActive == true)
{
// If we are already on a call return a busy response.
Log.LogWarning($"Busy response returned for incoming call request.");
uas.Reject(SIPResponseStatusCodesEnum.BusyHere, null);
}
else
{
var voipSession = new VoIPMediaSession(winAudio.ToMediaEndPoints());
voipSession.AcceptRtpFromAny = true;
var answerResult = await userAgent.Answer(uas, voipSession);
}
};
// At this point the call has been initiated and everything will be handled in an event handler.
Task.Run(async() =>
{
try
{
while (!exitCts.Token.WaitHandle.WaitOne(0))
{
var keyProps = Console.ReadKey();
if (keyProps.KeyChar == 'c')
{
if (!userAgent.IsCallActive)
{
var voipSession = new VoIPMediaSession(winAudio.ToMediaEndPoints());
voipSession.AcceptRtpFromAny = true;
bool callResult = await userAgent.Call(DEFAULT_DESTINATION_SIP_URI, SIP_USERNAME, SIP_PASSWORD, voipSession);
Log.LogInformation($"Call attempt {((callResult) ? "successfull" : "failed")}.");
}
else
{
Log.LogWarning("There is already an active call.");
}
}
else if (keyProps.KeyChar == 'h')
{
// Place call on/off hold.
if (userAgent.IsCallActive)
{
if (userAgent.IsOnLocalHold)
{
Log.LogInformation("Taking the remote call party off hold.");
(userAgent.MediaSession as VoIPMediaSession).TakeOffHold();
userAgent.TakeOffHold();
}
else
{
Log.LogInformation("Placing the remote call party on hold.");
await(userAgent.MediaSession as VoIPMediaSession).PutOnHold();
userAgent.PutOnHold();
}
}
else
{
Log.LogWarning("There is no active call to put on hold.");
}
}
else if (keyProps.KeyChar == 'H')
{
if (userAgent.IsCallActive)
{
Log.LogInformation("Hanging up call.");
userAgent.Hangup();
}
}
else if (keyProps.KeyChar == 't')
{
// Initiate a blind transfer to the remote call party.
if (userAgent.IsCallActive)
{
var transferURI = SIPURI.ParseSIPURI(TRANSFER_DESTINATION_SIP_URI);
bool result = await userAgent.BlindTransfer(transferURI, TimeSpan.FromSeconds(TRANSFER_TIMEOUT_SECONDS), exitCts.Token);
if (result)
{
// If the transfer was accepted the original call will already have been hungup.
// Wait a second for the transfer NOTIFY request to arrive.
await Task.Delay(1000);
exitCts.Cancel();
}
else
{
Log.LogWarning($"Transfer to {TRANSFER_DESTINATION_SIP_URI} failed.");
}
}
else
{
Log.LogWarning("There is no active call to transfer.");
}
}
else if (keyProps.KeyChar == 'q')
{
// Quit application.
exitCts.Cancel();
}
}
}
catch (Exception excp)
{
Log.LogError($"Exception Key Press listener. {excp.Message}.");
}
});
// Ctrl-c will gracefully exit the call at any point.
Console.CancelKeyPress += delegate(object sender, ConsoleCancelEventArgs e)
{
e.Cancel = true;
exitCts.Cancel();
};
// Wait for a signal saying the call failed, was cancelled with ctrl-c or completed.
exitCts.Token.WaitHandle.WaitOne();
#region Cleanup.
Log.LogInformation("Exiting...");
if (userAgent != null)
{
if (userAgent.IsCallActive)
{
Log.LogInformation($"Hanging up call to {userAgent?.CallDescriptor?.To}.");
userAgent.Hangup();
}
// Give the BYE or CANCEL request time to be transmitted.
Log.LogInformation("Waiting 1s for call to clean up...");
Task.Delay(1000).Wait();
}
if (sipTransport != null)
{
Log.LogInformation("Shutting down SIP transport...");
sipTransport.Shutdown();
}
#endregion
}
//private delegate void MediaSampleReadyDelegate(SDPMediaTypesEnum mediaType, uint duration, byte[] sample);
//private static event MediaSampleReadyDelegate OnMediaFromSIPSampleReady;
static void Main(string[] args)
{
Console.WriteLine("SIPSorcery SIP to WebRTC example.");
Console.WriteLine("Press ctrl-c to exit.");
// Plumbing code to facilitate a graceful exit.
CancellationTokenSource exitCts = new CancellationTokenSource(); // Cancellation token to stop the SIP transport and RTP stream.
Log = AddConsoleLogger();
// Start web socket.
Console.WriteLine("Starting web socket server...");
_webSocketServer = new WebSocketServer(IPAddress.Any, WEBSOCKET_PORT, true);
_webSocketServer.SslConfiguration.ServerCertificate = new X509Certificate2(WEBSOCKET_CERTIFICATE_PATH);
_webSocketServer.SslConfiguration.CheckCertificateRevocation = false;
//_webSocketServer.Log.Level = WebSocketSharp.LogLevel.Debug;
_webSocketServer.AddWebSocketService <SDPExchange>("/", (sdpExchanger) =>
{
sdpExchanger.WebSocketOpened += SendSDPOffer;
sdpExchanger.SDPAnswerReceived += SDPAnswerReceived;
});
_webSocketServer.Start();
Console.WriteLine($"Waiting for browser web socket connection to {_webSocketServer.Address}:{_webSocketServer.Port}...");
// Set up a default SIP transport.
var sipTransport = new SIPTransport();
sipTransport.AddSIPChannel(new SIPUDPChannel(new IPEndPoint(IPAddress.Any, SIP_LISTEN_PORT)));
//EnableTraceLogs(sipTransport);
RTPSession rtpSession = null;
// Create a SIP user agent to receive a call from a remote SIP client.
// Wire up event handlers for the different stages of the call.
var userAgent = new SIPUserAgent(sipTransport, null);
// We're only answering SIP calls, not placing them.
userAgent.OnCallHungup += (dialog) =>
{
Log.LogInformation($"Call hungup by remote party.");
exitCts.Cancel();
};
userAgent.ServerCallCancelled += (uas) => Log.LogInformation("Incoming call cancelled by caller.");
sipTransport.SIPTransportRequestReceived += async(localEndPoint, remoteEndPoint, sipRequest) =>
{
if (sipRequest.Header.From != null &&
sipRequest.Header.From.FromTag != null &&
sipRequest.Header.To != null &&
sipRequest.Header.To.ToTag != null)
{
// This is an in-dialog request that will be handled directly by a user agent instance.
}
else if (sipRequest.Method == SIPMethodsEnum.INVITE)
{
if (userAgent?.IsCallActive == true)
{
Log.LogWarning($"Busy response returned for incoming call request from {remoteEndPoint}: {sipRequest.StatusLine}.");
// If we are already on a call return a busy response.
UASInviteTransaction uasTransaction = new UASInviteTransaction(sipTransport, sipRequest, null);
SIPResponse busyResponse = SIPResponse.GetResponse(sipRequest, SIPResponseStatusCodesEnum.BusyHere, null);
uasTransaction.SendFinalResponse(busyResponse);
}
else
{
Log.LogInformation($"Incoming call request from {remoteEndPoint}: {sipRequest.StatusLine}.");
var incomingCall = userAgent.AcceptCall(sipRequest);
rtpSession = new RTPSession(false, false, false);
rtpSession.AcceptRtpFromAny = true;
MediaStreamTrack audioTrack = new MediaStreamTrack(SDPMediaTypesEnum.audio, false, new List <SDPMediaFormat> {
new SDPMediaFormat(SDPMediaFormatsEnum.PCMU)
});
rtpSession.addTrack(audioTrack);
await userAgent.Answer(incomingCall, rtpSession);
rtpSession.OnRtpPacketReceived += (ep, mediaType, rtpPacket) => ForwardMedia(mediaType, rtpPacket);
Log.LogInformation($"Answered incoming call from {sipRequest.Header.From.FriendlyDescription()} at {remoteEndPoint}.");
}
}
else
{
Log.LogDebug($"SIP {sipRequest.Method} request received but no processing has been set up for it, rejecting.");
SIPResponse notAllowedResponse = SIPResponse.GetResponse(sipRequest, SIPResponseStatusCodesEnum.MethodNotAllowed, null);
await sipTransport.SendResponseAsync(notAllowedResponse);
}
};
// Ctrl-c will gracefully exit the call at any point.
Console.CancelKeyPress += delegate(object sender, ConsoleCancelEventArgs e)
{
e.Cancel = true;
exitCts.Cancel();
};
// Wait for a signal saying the call failed, was cancelled with ctrl-c or completed.
exitCts.Token.WaitHandle.WaitOne();
#region Cleanup.
Log.LogInformation("Exiting...");
rtpSession?.Close("app exit");
if (userAgent != null)
{
if (userAgent.IsCallActive)
{
Log.LogInformation($"Hanging up call to {userAgent?.CallDescriptor?.To}.");
userAgent.Hangup();
}
// Give the BYE or CANCEL request time to be transmitted.
Log.LogInformation("Waiting 1s for call to clean up...");
Task.Delay(1000).Wait();
}
if (sipTransport != null)
{
Log.LogInformation("Shutting down SIP transport...");
sipTransport.Shutdown();
}
#endregion
}
static void Main()
{
Console.WriteLine("SIPSorcery call hold example.");
Console.WriteLine("Press ctrl-c to exit.");
// Plumbing code to facilitate a graceful exit.
CancellationTokenSource exitCts = new CancellationTokenSource(); // Cancellation token to stop the SIP transport and RTP stream.
AddConsoleLogger();
// Set up a default SIP transport.
var sipTransport = new SIPTransport();
sipTransport.AddSIPChannel(new SIPUDPChannel(new IPEndPoint(IPAddress.Any, SIP_LISTEN_PORT)));
EnableTraceLogs(sipTransport);
// Create two user agents. Each gets configured to answer an incoming call.
var userAgent1 = new SIPUserAgent(sipTransport, null);
var userAgent2 = new SIPUserAgent(sipTransport, null);
// Only one of the user agents can use the microphone and speaker. The one designated
// as the active agent gets the devices.
SIPUserAgent activeUserAgent = null;
RTPMediaSession activeRtpSession = null;
// Get the default speaker.
var(audioOutEvent, audioOutProvider) = GetAudioOutputDevice();
m_audioOutProvider = audioOutProvider;
WaveInEvent waveInEvent = GetAudioInputDevice();
userAgent1.OnCallHungup += () => Log.LogInformation($"UA1: Call hungup by remote party.");
userAgent1.ServerCallCancelled += (uas) => Log.LogInformation("UA1: Incoming call cancelled by caller.");
userAgent2.OnCallHungup += () => Log.LogInformation($"UA2: Call hungup by remote party.");
userAgent2.ServerCallCancelled += (uas) => Log.LogInformation("UA2: Incoming call cancelled by caller.");
userAgent2.OnTransferNotify += (sipFrag) =>
{
if (!string.IsNullOrEmpty(sipFrag))
{
Log.LogInformation($"UA2: Transfer status update: {sipFrag.Trim()}.");
if (sipFrag?.Contains("SIP/2.0 200") == true)
{
// The transfer attempt got a succesful answer. Can hangup the call.
userAgent2.Hangup();
exitCts.Cancel();
}
}
};
sipTransport.SIPTransportRequestReceived += (locelEndPoint, remoteEndPoint, sipRequest) =>
{
if (sipRequest.Header.From != null &&
sipRequest.Header.From.FromTag != null &&
sipRequest.Header.To != null &&
sipRequest.Header.To.ToTag != null)
{
// This is an in-dialog request that will be handled directly by a user agent instance.
}
else if (sipRequest.Method == SIPMethodsEnum.INVITE)
{
if (!userAgent1.IsCallActive)
{
Log.LogInformation($"UA1: Incoming call request from {remoteEndPoint}: {sipRequest.StatusLine}.");
var incomingCall = userAgent1.AcceptCall(sipRequest);
var rtpMediaSession = new RTPMediaSession(SDPMediaTypesEnum.audio, new SDPMediaFormat(SDPMediaFormatsEnum.PCMU), AddressFamily.InterNetwork);
rtpMediaSession.RemotePutOnHold += () => Log.LogInformation("UA1: Remote call party has placed us on hold.");
rtpMediaSession.RemoteTookOffHold += () => Log.LogInformation("UA1: Remote call party took us off hold.");
userAgent1.Answer(incomingCall, rtpMediaSession)
.ContinueWith(task =>
{
activeUserAgent = userAgent1;
activeRtpSession = rtpMediaSession;
activeRtpSession.OnRtpPacketReceived += PlaySample;
waveInEvent.StartRecording();
Log.LogInformation($"UA1: Answered incoming call from {sipRequest.Header.From.FriendlyDescription()} at {remoteEndPoint}.");
}, exitCts.Token);
}
else if (!userAgent2.IsCallActive)
{
Log.LogInformation($"UA2: Incoming call request from {remoteEndPoint}: {sipRequest.StatusLine}.");
var incomingCall = userAgent2.AcceptCall(sipRequest);
var rtpMediaSession = new RTPMediaSession(SDPMediaTypesEnum.audio, new SDPMediaFormat(SDPMediaFormatsEnum.PCMU), AddressFamily.InterNetwork);
rtpMediaSession.RemotePutOnHold += () => Log.LogInformation("UA2: Remote call party has placed us on hold.");
rtpMediaSession.RemoteTookOffHold += () => Log.LogInformation("UA2: Remote call party took us off hold.");
userAgent2.Answer(incomingCall, rtpMediaSession)
.ContinueWith(task =>
{
activeRtpSession.OnRtpPacketReceived -= PlaySample;
activeUserAgent = userAgent2;
activeRtpSession = rtpMediaSession;
activeRtpSession.PutOnHold();
activeRtpSession.OnRtpPacketReceived += PlaySample;
Log.LogInformation($"UA2: Answered incoming call from {sipRequest.Header.From.FriendlyDescription()} at {remoteEndPoint}.");
}, exitCts.Token);
}
else
{
// If both user agents are already on a call return a busy response.
Log.LogWarning($"Busy response returned for incoming call request from {remoteEndPoint}: {sipRequest.StatusLine}.");
UASInviteTransaction uasTransaction = new UASInviteTransaction(sipTransport, sipRequest, null);
SIPResponse busyResponse = SIPResponse.GetResponse(sipRequest, SIPResponseStatusCodesEnum.BusyHere, null);
uasTransaction.SendFinalResponse(busyResponse);
}
}
else
{
Log.LogDebug($"SIP {sipRequest.Method} request received but no processing has been set up for it, rejecting.");
SIPResponse notAllowedResponse = SIPResponse.GetResponse(sipRequest, SIPResponseStatusCodesEnum.MethodNotAllowed, null);
return(sipTransport.SendResponseAsync(notAllowedResponse));
}
return(Task.FromResult(0));
};
// Wire up the RTP send session to the audio input device.
uint rtpSendTimestamp = 0;
waveInEvent.DataAvailable += (object sender, WaveInEventArgs args) =>
{
byte[] sample = new byte[args.Buffer.Length / 2];
int sampleIndex = 0;
for (int index = 0; index < args.BytesRecorded; index += 2)
{
var ulawByte = NAudio.Codecs.MuLawEncoder.LinearToMuLawSample(BitConverter.ToInt16(args.Buffer, index));
sample[sampleIndex++] = ulawByte;
}
if (activeRtpSession != null)
{
activeRtpSession.SendAudioFrame(rtpSendTimestamp, (int)SDPMediaFormatsEnum.PCMU, sample);
rtpSendTimestamp += (uint)sample.Length;
}
};
// At this point the call has been initiated and everything will be handled in an event handler.
Task.Run(async() =>
{
try
{
while (!exitCts.Token.WaitHandle.WaitOne(0))
{
var keyProps = Console.ReadKey();
if (keyProps.KeyChar == 't')
{
if (userAgent1.IsCallActive && userAgent2.IsCallActive)
{
bool result = await userAgent2.AttendedTransfer(userAgent1.Dialogue, TimeSpan.FromSeconds(TRANSFER_TIMEOUT_SECONDS), exitCts.Token);
if (!result)
{
Log.LogWarning($"Attended transfer failed.");
}
}
else
{
Log.LogWarning("There need to be two active calls before the attended transfer can occur.");
}
}
else if (keyProps.KeyChar == 'q')
{
// Quit application.
exitCts.Cancel();
}
}
}
catch (Exception excp)
{
SIPSorcery.Sys.Log.Logger.LogError($"Exception Key Press listener. {excp.Message}.");
}
});
// Ctrl-c will gracefully exit the call at any point.
Console.CancelKeyPress += delegate(object sender, ConsoleCancelEventArgs e)
{
e.Cancel = true;
exitCts.Cancel();
};
// Wait for a signal saying the call failed, was cancelled with ctrl-c or completed.
exitCts.Token.WaitHandle.WaitOne();
#region Cleanup.
Log.LogInformation("Exiting...");
userAgent1?.Hangup();
userAgent2?.Hangup();
waveInEvent?.StopRecording();
audioOutEvent?.Stop();
// Give any BYE or CANCEL requests time to be transmitted.
Log.LogInformation("Waiting 1s for calls to be cleaned up...");
Task.Delay(1000).Wait();
SIPSorcery.Net.DNSManager.Stop();
if (sipTransport != null)
{
Log.LogInformation("Shutting down SIP transport...");
sipTransport.Shutdown();
}
#endregion
}
static void Main()
{
Console.WriteLine("SIPSorcery Call Hold and Blind Transfer example.");
Console.WriteLine("Press 'c' to initiate a call to the default destination.");
Console.WriteLine("Press 'h' to place an established call on and off hold.");
Console.WriteLine("Press 'H' to hangup an established call.");
Console.WriteLine("Press 't' to request a blind transfer on an established call.");
Console.WriteLine("Press 'q' or ctrl-c to exit.");
// Plumbing code to facilitate a graceful exit.
CancellationTokenSource exitCts = new CancellationTokenSource(); // Cancellation token to stop the SIP transport and RTP stream.
AddConsoleLogger();
// Set up a default SIP transport.
var sipTransport = new SIPTransport();
sipTransport.AddSIPChannel(new SIPUDPChannel(new IPEndPoint(IPAddress.Any, SIP_LISTEN_PORT)));
Console.WriteLine($"Listening for incoming calls on: {sipTransport.GetSIPChannels().First().ListeningEndPoint}.");
EnableTraceLogs(sipTransport);
_currentDir = Path.GetDirectoryName(System.Reflection.Assembly.GetExecutingAssembly().Location);
RtpAVSession rtpAVSession = null;
// Create a client/server user agent to place a call to a remote SIP server along with event handlers for the different stages of the call.
var userAgent = new SIPUserAgent(sipTransport, null);
userAgent.RemotePutOnHold += () => Log.LogInformation("Remote call party has placed us on hold.");
userAgent.RemoteTookOffHold += () => Log.LogInformation("Remote call party took us off hold.");
sipTransport.SIPTransportRequestReceived += async(localEndPoint, remoteEndPoint, sipRequest) =>
{
if (sipRequest.Header.From != null &&
sipRequest.Header.From.FromTag != null &&
sipRequest.Header.To != null &&
sipRequest.Header.To.ToTag != null)
{
// This is an in-dialog request that will be handled directly by a user agent instance.
}
else if (sipRequest.Method == SIPMethodsEnum.INVITE)
{
if (userAgent?.IsCallActive == true)
{
Log.LogWarning($"Busy response returned for incoming call request from {remoteEndPoint}: {sipRequest.StatusLine}.");
// If we are already on a call return a busy response.
UASInviteTransaction uasTransaction = new UASInviteTransaction(sipTransport, sipRequest, null);
SIPResponse busyResponse = SIPResponse.GetResponse(sipRequest, SIPResponseStatusCodesEnum.BusyHere, null);
uasTransaction.SendFinalResponse(busyResponse);
}
else
{
Log.LogInformation($"Incoming call request from {remoteEndPoint}: {sipRequest.StatusLine}.");
var incomingCall = userAgent.AcceptCall(sipRequest);
rtpAVSession = new RtpAVSession(new AudioOptions {
AudioSource = AudioSourcesEnum.CaptureDevice
}, null);
await userAgent.Answer(incomingCall, rtpAVSession);
Log.LogInformation($"Answered incoming call from {sipRequest.Header.From.FriendlyDescription()} at {remoteEndPoint}.");
}
}
else
{
Log.LogDebug($"SIP {sipRequest.Method} request received but no processing has been set up for it, rejecting.");
SIPResponse notAllowedResponse = SIPResponse.GetResponse(sipRequest, SIPResponseStatusCodesEnum.MethodNotAllowed, null);
await sipTransport.SendResponseAsync(notAllowedResponse);
}
};
// At this point the call has been initiated and everything will be handled in an event handler.
Task.Run(async() =>
{
try
{
while (!exitCts.Token.WaitHandle.WaitOne(0))
{
var keyProps = Console.ReadKey();
if (keyProps.KeyChar == 'c')
{
if (!userAgent.IsCallActive)
{
rtpAVSession = new RtpAVSession(new AudioOptions {
AudioSource = AudioSourcesEnum.CaptureDevice
}, null);
bool callResult = await userAgent.Call(DEFAULT_DESTINATION_SIP_URI, SIP_USERNAME, SIP_PASSWORD, rtpAVSession);
Log.LogInformation($"Call attempt {((callResult) ? "successfull" : "failed")}.");
}
else
{
Log.LogWarning("There is already an active call.");
}
}
else if (keyProps.KeyChar == 'h')
{
// Place call on/off hold.
if (userAgent.IsCallActive)
{
if (userAgent.IsOnLocalHold)
{
Log.LogInformation("Taking the remote call party off hold.");
userAgent.TakeOffHold();
await(userAgent.MediaSession as RtpAVSession).SetSources(new AudioOptions {
AudioSource = AudioSourcesEnum.CaptureDevice
}, null);
}
else
{
Log.LogInformation("Placing the remote call party on hold.");
userAgent.PutOnHold();
await(userAgent.MediaSession as RtpAVSession).SetSources(new AudioOptions
{
AudioSource = AudioSourcesEnum.Music,
SourceFiles = new Dictionary <SDPMediaFormatsEnum, string>
{
{ SDPMediaFormatsEnum.PCMU, _currentDir + "/" + AUDIO_FILE_PCMU }
}
}, null);
}
}
else
{
Log.LogWarning("There is no active call to put on hold.");
}
}
else if (keyProps.KeyChar == 'H')
{
if (userAgent.IsCallActive)
{
Log.LogInformation("Hanging up call.");
userAgent.Hangup();
}
}
else if (keyProps.KeyChar == 't')
{
// Initiate a blind transfer to the remote call party.
if (userAgent.IsCallActive)
{
var transferURI = SIPURI.ParseSIPURI(TRANSFER_DESTINATION_SIP_URI);
bool result = await userAgent.BlindTransfer(transferURI, TimeSpan.FromSeconds(TRANSFER_TIMEOUT_SECONDS), exitCts.Token);
if (result)
{
// If the transfer was accepted the original call will already have been hungup.
// Wait a second for the transfer NOTIFY request to arrive.
await Task.Delay(1000);
exitCts.Cancel();
}
else
{
Log.LogWarning($"Transfer to {TRANSFER_DESTINATION_SIP_URI} failed.");
}
}
else
{
Log.LogWarning("There is no active call to transfer.");
}
}
else if (keyProps.KeyChar == 'q')
{
// Quit application.
exitCts.Cancel();
}
}
}
catch (Exception excp)
{
SIPSorcery.Sys.Log.Logger.LogError($"Exception Key Press listener. {excp.Message}.");
}
});
// Ctrl-c will gracefully exit the call at any point.
Console.CancelKeyPress += delegate(object sender, ConsoleCancelEventArgs e)
{
e.Cancel = true;
exitCts.Cancel();
};
// Wait for a signal saying the call failed, was cancelled with ctrl-c or completed.
exitCts.Token.WaitHandle.WaitOne();
#region Cleanup.
Log.LogInformation("Exiting...");
rtpAVSession?.Close("app exit");
if (userAgent != null)
{
if (userAgent.IsCallActive)
{
Log.LogInformation($"Hanging up call to {userAgent?.CallDescriptor?.To}.");
userAgent.Hangup();
}
// Give the BYE or CANCEL request time to be transmitted.
Log.LogInformation("Waiting 1s for call to clean up...");
Task.Delay(1000).Wait();
}
SIPSorcery.Net.DNSManager.Stop();
if (sipTransport != null)
{
Log.LogInformation("Shutting down SIP transport...");
sipTransport.Shutdown();
}
#endregion
}
static void Main()
{
Console.WriteLine("SIPSorcery Attended Transfer example.");
Console.WriteLine("Press 'c' to place a call to the default destination.");
Console.WriteLine("Place two simultaneous SIP calls to this program and then press 't'.");
Console.WriteLine("Press 'q' or ctrl-c to exit.");
// Plumbing code to facilitate a graceful exit.
CancellationTokenSource exitCts = new CancellationTokenSource(); // Cancellation token to stop the SIP transport and RTP stream.
AddConsoleLogger();
// Set up a default SIP transport.
var sipTransport = new SIPTransport();
sipTransport.AddSIPChannel(new SIPUDPChannel(new IPEndPoint(IPAddress.Any, SIP_LISTEN_PORT)));
//EnableTraceLogs(sipTransport);
// Create two user agents. Each gets configured to answer an incoming call.
var userAgent1 = new SIPUserAgent(sipTransport, null);
var userAgent2 = new SIPUserAgent(sipTransport, null);
userAgent1.OnCallHungup += (dialog) => Log.LogInformation($"UA1: Call hungup by remote party.");
userAgent1.ServerCallCancelled += (uas) => Log.LogInformation("UA1: Incoming call cancelled by caller.");
userAgent2.OnCallHungup += (dialog) => Log.LogInformation($"UA2: Call hungup by remote party.");
userAgent2.ServerCallCancelled += (uas) => Log.LogInformation("UA2: Incoming call cancelled by caller.");
userAgent2.OnTransferNotify += (sipFrag) =>
{
if (!string.IsNullOrEmpty(sipFrag))
{
Log.LogInformation($"UA2: Transfer status update: {sipFrag.Trim()}.");
if (sipFrag?.Contains("SIP/2.0 200") == true)
{
// The transfer attempt got a successful answer. Can hangup the call.
userAgent2.Hangup();
exitCts.Cancel();
}
}
};
sipTransport.SIPTransportRequestReceived += async(localEndPoint, remoteEndPoint, sipRequest) =>
{
if (sipRequest.Header.From != null &&
sipRequest.Header.From.FromTag != null &&
sipRequest.Header.To != null &&
sipRequest.Header.To.ToTag != null)
{
// This is an in-dialog request that will be handled directly by a user agent instance.
}
else if (sipRequest.Method == SIPMethodsEnum.INVITE)
{
if (!userAgent1.IsCallActive || !userAgent2.IsCallActive)
{
SIPUserAgent activeAgent = (!userAgent1.IsCallActive) ? userAgent1 : userAgent2;
string agentDesc = (!userAgent1.IsCallActive) ? "UA1" : "UA2";
Log.LogInformation($"{agentDesc}: Incoming call request from {remoteEndPoint}: {sipRequest.StatusLine}.");
var incomingCall = activeAgent.AcceptCall(sipRequest);
var rtpAVSession = new RtpAVSession(new AudioOptions {
AudioSource = AudioSourcesEnum.CaptureDevice
}, null);
await activeAgent.Answer(incomingCall, rtpAVSession);
Log.LogInformation($"{agentDesc}: Answered incoming call from {sipRequest.Header.From.FriendlyDescription()} at {remoteEndPoint}.");
}
else
{
// If both user agents are already on a call return a busy response.
Log.LogWarning($"Busy response returned for incoming call request from {remoteEndPoint}: {sipRequest.StatusLine}.");
UASInviteTransaction uasTransaction = new UASInviteTransaction(sipTransport, sipRequest, null);
SIPResponse busyResponse = SIPResponse.GetResponse(sipRequest, SIPResponseStatusCodesEnum.BusyHere, null);
uasTransaction.SendFinalResponse(busyResponse);
}
}
else
{
Log.LogDebug($"SIP {sipRequest.Method} request received but no processing has been set up for it, rejecting.");
SIPResponse notAllowedResponse = SIPResponse.GetResponse(sipRequest, SIPResponseStatusCodesEnum.MethodNotAllowed, null);
await sipTransport.SendResponseAsync(notAllowedResponse);
}
};
// At this point the call has been initiated and everything will be handled in an event handler.
Task.Run(async() =>
{
try
{
while (!exitCts.Token.WaitHandle.WaitOne(0))
{
var keyProps = Console.ReadKey();
if (keyProps.KeyChar == 'c')
{
// Place an outgoing call using the first free user agent.
SIPUserAgent freeAgent = (!userAgent1.IsCallActive) ? userAgent1 : (!userAgent2.IsCallActive) ? userAgent2 : null;
if (freeAgent != null)
{
var rtpAVSession = new RtpAVSession(new AudioOptions {
AudioSource = AudioSourcesEnum.CaptureDevice
}, null);
bool callResult = await freeAgent.Call(DEFAULT_DESTINATION_SIP_URI, null, null, rtpAVSession);
Log.LogInformation($"Call attempt {((callResult) ? "successfull" : "failed")}.");
}
else
{
Log.LogWarning("Both user agents are already on calls.");
}
}
if (keyProps.KeyChar == 't')
{
// Initiate the attended transfer.
if (userAgent1.IsCallActive && userAgent2.IsCallActive)
{
bool result = await userAgent2.AttendedTransfer(userAgent1.Dialogue, TimeSpan.FromSeconds(TRANSFER_TIMEOUT_SECONDS), exitCts.Token);
if (!result)
{
Log.LogWarning($"Attended transfer failed.");
}
}
else
{
Log.LogWarning("There need to be two active calls before the attended transfer can occur.");
}
}
else if (keyProps.KeyChar == 'q')
{
// Quit application.
exitCts.Cancel();
}
}
}
catch (Exception excp)
{
SIPSorcery.Sys.Log.Logger.LogError($"Exception Key Press listener. {excp.Message}.");
}
});
// Ctrl-c will gracefully exit the call at any point.
Console.CancelKeyPress += delegate(object sender, ConsoleCancelEventArgs e)
{
e.Cancel = true;
exitCts.Cancel();
};
// Wait for a signal saying the call failed, was cancelled with ctrl-c or completed.
exitCts.Token.WaitHandle.WaitOne();
#region Cleanup.
Log.LogInformation("Exiting...");
userAgent1?.Hangup();
userAgent2?.Hangup();
// Give any BYE or CANCEL requests time to be transmitted.
Log.LogInformation("Waiting 1s for calls to be cleaned up...");
Task.Delay(1000).Wait();
if (sipTransport != null)
{
Log.LogInformation("Shutting down SIP transport...");
sipTransport.Shutdown();
}
#endregion
}
static void Main()
{
Console.WriteLine("SIPSorcery SIP to WebRTC example.");
Console.WriteLine("Press ctrl-c to exit.");
// Plumbing code to facilitate a graceful exit.
CancellationTokenSource exitCts = new CancellationTokenSource(); // Cancellation token to stop the SIP transport and RTP stream.
Log = AddConsoleLogger();
//EnableTraceLogs(sipTransport);
// Start web socket.
Console.WriteLine("Starting web socket server...");
var webSocketServer = new WebSocketServer(IPAddress.Any, WEBSOCKET_PORT);
webSocketServer.AddWebSocketService <WebRTCWebSocketPeer>("/", (peer) => peer.CreatePeerConnection = CreatePeerConnection);
webSocketServer.Start();
// Set up a default SIP transport.
var sipTransport = new SIPTransport();
sipTransport.AddSIPChannel(new SIPUDPChannel(new IPEndPoint(IPAddress.Any, SIP_LISTEN_PORT)));
// Create a SIP user agent to receive a call from a remote SIP client.
// Wire up event handlers for the different stages of the call.
var userAgent = new SIPUserAgent(sipTransport, null, true);
// We're only answering SIP calls, not placing them.
userAgent.OnCallHungup += (dialog) =>
{
Log.LogInformation($"Call hungup by remote party.");
exitCts.Cancel();
};
userAgent.ServerCallCancelled += (uas) => Log.LogInformation("Incoming call cancelled by caller.");
userAgent.OnIncomingCall += async(ua, req) =>
{
Log.LogInformation($"Incoming call request from {req.RemoteSIPEndPoint}: {req.StatusLine}.");
var incomingCall = userAgent.AcceptCall(req);
var rtpSession = new RTPSession(false, false, false);
rtpSession.AcceptRtpFromAny = true;
MediaStreamTrack audioTrack = new MediaStreamTrack(SDPMediaTypesEnum.audio, false,
new List <SDPAudioVideoMediaFormat> {
new SDPAudioVideoMediaFormat(SDPWellKnownMediaFormatsEnum.PCMU)
});
rtpSession.addTrack(audioTrack);
await userAgent.Answer(incomingCall, rtpSession);
rtpSession.OnRtpPacketReceived += ForwardMediaToPeerConnection;
Log.LogInformation($"Answered incoming call from {req.Header.From.FriendlyDescription()} at {req.RemoteSIPEndPoint}.");
_rtpSession = rtpSession;
};
Console.WriteLine($"Waiting for browser web socket connection to {webSocketServer.Address}:{webSocketServer.Port}...");
var contactURI = new SIPURI(SIPSchemesEnum.sip, sipTransport.GetSIPChannels().First().ListeningSIPEndPoint);
Console.WriteLine($"Waiting for incoming SIP call to {contactURI}.");
// Ctrl-c will gracefully exit the call at any point.
Console.CancelKeyPress += delegate(object sender, ConsoleCancelEventArgs e)
{
e.Cancel = true;
exitCts.Cancel();
};
// Wait for a signal saying the call failed, was cancelled with ctrl-c or completed.
exitCts.Token.WaitHandle.WaitOne();
#region Cleanup.
Log.LogInformation("Exiting...");
_rtpSession?.Close("app exit");
if (userAgent != null)
{
if (userAgent.IsCallActive)
{
Log.LogInformation($"Hanging up call to {userAgent?.CallDescriptor?.To}.");
userAgent.Hangup();
}
// Give the BYE or CANCEL request time to be transmitted.
Log.LogInformation("Waiting 1s for call to clean up...");
Task.Delay(1000).Wait();
}
if (sipTransport != null)
{
Log.LogInformation("Shutting down SIP transport...");
sipTransport.Shutdown();
}
#endregion
}
static void Main()
{
Console.WriteLine("SIPSorcery call hold example.");
Console.WriteLine("Press ctrl-c to exit.");
// Plumbing code to facilitate a graceful exit.
CancellationTokenSource exitCts = new CancellationTokenSource(); // Cancellation token to stop the SIP transport and RTP stream.
bool isCallHungup = false;
bool hasCallFailed = false;
AddConsoleLogger();
// Set up a default SIP transport.
var sipTransport = new SIPTransport();
sipTransport.AddSIPChannel(new SIPUDPChannel(new IPEndPoint(IPAddress.Any, SIP_LISTEN_PORT)));
//EnableTraceLogs(sipTransport);
// Get the default speaker.
var(audioOutEvent, audioOutProvider) = GetAudioOutputDevice();
WaveInEvent waveInEvent = GetAudioInputDevice();
RTPMediaSession RtpMediaSession = null;
// Create a client/server user agent to place a call to a remote SIP server along with event handlers for the different stages of the call.
var userAgent = new SIPUserAgent(sipTransport, null);
userAgent.ClientCallTrying += (uac, resp) => Log.LogInformation($"{uac.CallDescriptor.To} Trying: {resp.StatusCode} {resp.ReasonPhrase}.");
userAgent.ClientCallRinging += (uac, resp) => Log.LogInformation($"{uac.CallDescriptor.To} Ringing: {resp.StatusCode} {resp.ReasonPhrase}.");
userAgent.ClientCallFailed += (uac, err) =>
{
Log.LogWarning($"{uac.CallDescriptor.To} Failed: {err}");
hasCallFailed = true;
exitCts.Cancel();
};
userAgent.ClientCallAnswered += (uac, resp) =>
{
if (resp.Status == SIPResponseStatusCodesEnum.Ok)
{
Log.LogInformation($"{uac.CallDescriptor.To} Answered: {resp.StatusCode} {resp.ReasonPhrase}.");
PlayRemoteMedia(RtpMediaSession, audioOutProvider);
}
else
{
Log.LogWarning($"{uac.CallDescriptor.To} Answered: {resp.StatusCode} {resp.ReasonPhrase}.");
hasCallFailed = true;
exitCts.Cancel();
}
};
userAgent.OnCallHungup += () =>
{
Log.LogInformation($"Call hungup by remote party.");
exitCts.Cancel();
};
userAgent.ServerCallCancelled += (uas) => Log.LogInformation("Incoming call cancelled by caller.");
sipTransport.SIPTransportRequestReceived += async(localEndPoint, remoteEndPoint, sipRequest) =>
{
if (sipRequest.Header.From != null &&
sipRequest.Header.From.FromTag != null &&
sipRequest.Header.To != null &&
sipRequest.Header.To.ToTag != null)
{
// This is an in-dialog request that will be handled directly by a user agent instance.
}
else if (sipRequest.Method == SIPMethodsEnum.INVITE)
{
if (userAgent?.IsCallActive == true)
{
Log.LogWarning($"Busy response returned for incoming call request from {remoteEndPoint}: {sipRequest.StatusLine}.");
// If we are already on a call return a busy response.
UASInviteTransaction uasTransaction = new UASInviteTransaction(sipTransport, sipRequest, null);
SIPResponse busyResponse = SIPResponse.GetResponse(sipRequest, SIPResponseStatusCodesEnum.BusyHere, null);
uasTransaction.SendFinalResponse(busyResponse);
}
else
{
Log.LogInformation($"Incoming call request from {remoteEndPoint}: {sipRequest.StatusLine}.");
var incomingCall = userAgent.AcceptCall(sipRequest);
RtpMediaSession = new RTPMediaSession(SDPMediaTypesEnum.audio, (int)SDPMediaFormatsEnum.PCMU, AddressFamily.InterNetwork);
RtpMediaSession.RemotePutOnHold += () => Log.LogInformation("Remote call party has placed us on hold.");
RtpMediaSession.RemoteTookOffHold += () => Log.LogInformation("Remote call party took us off hold.");
await userAgent.Answer(incomingCall, RtpMediaSession);
PlayRemoteMedia(RtpMediaSession, audioOutProvider);
waveInEvent.StartRecording();
Log.LogInformation($"Answered incoming call from {sipRequest.Header.From.FriendlyDescription()} at {remoteEndPoint}.");
}
}
else
{
Log.LogDebug($"SIP {sipRequest.Method} request received but no processing has been set up for it, rejecting.");
SIPResponse notAllowedResponse = SIPResponse.GetResponse(sipRequest, SIPResponseStatusCodesEnum.MethodNotAllowed, null);
await sipTransport.SendResponseAsync(notAllowedResponse);
}
};
// Wire up the RTP send session to the audio output device.
uint rtpSendTimestamp = 0;
waveInEvent.DataAvailable += (object sender, WaveInEventArgs args) =>
{
byte[] sample = new byte[args.Buffer.Length / 2];
int sampleIndex = 0;
for (int index = 0; index < args.BytesRecorded; index += 2)
{
var ulawByte = NAudio.Codecs.MuLawEncoder.LinearToMuLawSample(BitConverter.ToInt16(args.Buffer, index));
sample[sampleIndex++] = ulawByte;
}
if (RtpMediaSession != null)
{
RtpMediaSession.SendAudioFrame(rtpSendTimestamp, sample);
rtpSendTimestamp += (uint)(8000 / waveInEvent.BufferMilliseconds);
}
};
// At this point the call has been initiated and everything will be handled in an event handler.
Task.Run(async() =>
{
try
{
while (!exitCts.Token.WaitHandle.WaitOne(0))
{
var keyProps = Console.ReadKey();
if (keyProps.KeyChar == 'c')
{
if (!userAgent.IsCallActive)
{
RtpMediaSession = new RTPMediaSession(SDPMediaTypesEnum.audio, (int)SDPMediaFormatsEnum.PCMU, AddressFamily.InterNetwork);
RtpMediaSession.RemotePutOnHold += () => Log.LogInformation("Remote call party has placed us on hold.");
RtpMediaSession.RemoteTookOffHold += () => Log.LogInformation("Remote call party took us off hold.");
var callDescriptor = GetCallDescriptor(DEFAULT_DESTINATION_SIP_URI);
await userAgent.InitiateCall(callDescriptor, RtpMediaSession);
}
else
{
Log.LogWarning("There is already an active call.");
}
}
else if (keyProps.KeyChar == 'h')
{
// Place call on/off hold.
if (userAgent.IsCallActive)
{
if (RtpMediaSession.LocalOnHold)
{
Log.LogInformation("Taking the remote call party off hold.");
RtpMediaSession.TakeOffHold();
}
else
{
Log.LogInformation("Placing the remote call party on hold.");
RtpMediaSession.PutOnHold();
}
}
else
{
Log.LogWarning("There is no active call to put on hold.");
}
}
else if (keyProps.KeyChar == 't')
{
if (userAgent.IsCallActive)
{
var transferURI = SIPURI.ParseSIPURI(TRANSFER_DESTINATION_SIP_URI);
bool result = await userAgent.BlindTransfer(transferURI, TimeSpan.FromSeconds(TRANSFER_TIMEOUT_SECONDS), exitCts.Token);
if (result)
{
// If the transfer was accepted the original call will already have been hungup.
// Wait a second for the transfer NOTIFY request to arrive.
await Task.Delay(1000);
exitCts.Cancel();
}
else
{
Log.LogWarning($"Transfer to {TRANSFER_DESTINATION_SIP_URI} failed.");
}
}
else
{
Log.LogWarning("There is no active call to transfer.");
}
}
else if (keyProps.KeyChar == 'q')
{
// Quit application.
exitCts.Cancel();
}
}
}
catch (Exception excp)
{
SIPSorcery.Sys.Log.Logger.LogError($"Exception Key Press listener. {excp.Message}.");
}
});
// Ctrl-c will gracefully exit the call at any point.
Console.CancelKeyPress += delegate(object sender, ConsoleCancelEventArgs e)
{
e.Cancel = true;
exitCts.Cancel();
};
// Wait for a signal saying the call failed, was cancelled with ctrl-c or completed.
exitCts.Token.WaitHandle.WaitOne();
#region Cleanup.
Log.LogInformation("Exiting...");
RtpMediaSession?.Close();
waveInEvent?.StopRecording();
audioOutEvent?.Stop();
if (!isCallHungup && userAgent != null)
{
if (userAgent.IsCallActive)
{
Log.LogInformation($"Hanging up call to {userAgent?.CallDescriptor?.To}.");
userAgent.Hangup();
}
else if (!hasCallFailed)
{
Log.LogInformation($"Cancelling call to {userAgent?.CallDescriptor?.To}.");
userAgent.Cancel();
}
// Give the BYE or CANCEL request time to be transmitted.
Log.LogInformation("Waiting 1s for call to clean up...");
Task.Delay(1000).Wait();
}
SIPSorcery.Net.DNSManager.Stop();
if (sipTransport != null)
{
Log.LogInformation("Shutting down SIP transport...");
sipTransport.Shutdown();
}
#endregion
}
/// <summary>
/// Accepts an incoming call. This is the first step in answering a call.
/// From this point the call can still be rejected, redirected or answered.
/// </summary>
/// <param name="sipRequest">The SIP request containing the incoming call request.</param>
public void Accept(SIPRequest sipRequest)
{
m_pendingIncomingCall = m_userAgent.AcceptCall(sipRequest);
}
static async Task Main()
{
Console.WriteLine("SIPSorcery sip.js Demo");
AddConsoleLogger();
var sipTransport = new SIPTransport();
EnableTraceLogs(sipTransport);
var sipChannel = new SIPWebSocketChannel(IPAddress.Loopback, 80);
var wssCertificate = new System.Security.Cryptography.X509Certificates.X509Certificate2("localhost.pfx");
var sipChannelSecure = new SIPWebSocketChannel(IPAddress.Loopback, 443, wssCertificate);
sipTransport.AddSIPChannel(sipChannel);
sipTransport.AddSIPChannel(sipChannelSecure);
var userAgent = new SIPUserAgent(sipTransport, null, true);
userAgent.OnIncomingCall += async(ua, req) =>
{
Log.LogDebug($"Auto-answering incoming call from {req.Header.From}.");
var uas = userAgent.AcceptCall(req);
RTCConfiguration pcConfiguration = new RTCConfiguration
{
certificates = new List <RTCCertificate>
{
new RTCCertificate
{
X_CertificatePath = DTLS_CERTIFICATE_PATH,
X_KeyPath = DTLS_KEY_PATH,
X_Fingerprint = DTLS_CERTIFICATE_FINGERPRINT
}
},
//X_RemoteSignallingAddress = context.UserEndPoint.Address,
//iceServers = new List<RTCIceServer> { new RTCIceServer { urls = SIPSORCERY_STUN_SERVER } }
};
var peerConnection = new RTCPeerConnection(pcConfiguration);
var dtls = new DtlsHandshake(DTLS_CERTIFICATE_PATH, DTLS_KEY_PATH);
peerConnection.OnTimeout += (mediaType) =>
{
peerConnection.Close("remote timeout");
};
peerConnection.oniceconnectionstatechange += async(state) =>
{
Log.LogDebug($"ICE connection state change to {state}.");
if (state == RTCIceConnectionState.connected)
{
var remoteEndPoint = peerConnection.AudioDestinationEndPoint;
Log.LogInformation($"ICE connected to remote end point {remoteEndPoint}.");
await Task.Run(() => DoDtlsHandshake(peerConnection, dtls))
.ContinueWith((dtlsResult) =>
{
Log.LogDebug($"dtls handshake result {dtlsResult.Result}.");
if (dtlsResult.Result)
{
var remoteEP = peerConnection.AudioDestinationEndPoint;
peerConnection.SetDestination(SDPMediaTypesEnum.audio, remoteEP, remoteEP);
}
else
{
dtls.Shutdown();
peerConnection.Close("dtls handshake failed.");
}
});
}
};
peerConnection.onconnectionstatechange += (state) =>
{
if (state == RTCPeerConnectionState.connected)
{
var remoteEP = peerConnection.AudioDestinationEndPoint;
Log.LogDebug($"DTLS connected on {remoteEP}.");
peerConnection.SetDestination(SDPMediaTypesEnum.audio, remoteEP, remoteEP);
peerConnection.SetDestination(SDPMediaTypesEnum.video, remoteEP, remoteEP);
peerConnection.OnReceiveReport += RtpSession_OnReceiveReport;
peerConnection.OnSendReport += RtpSession_OnSendReport;
// peerConnection.OnRtpPacketReceived += OnRtpPacketReceived;
}
};
MediaStreamTrack audioTrack = new MediaStreamTrack(SDPMediaTypesEnum.audio, false, new List <SDPMediaFormat> {
new SDPMediaFormat(SDPMediaFormatsEnum.PCMU)
}, MediaStreamStatusEnum.SendRecv);
peerConnection.addTrack(audioTrack);
//MediaStreamTrack videoTrack = new MediaStreamTrack("1", SDPMediaTypesEnum.video, false, new List<SDPMediaFormat> { new SDPMediaFormat(SDPMediaFormatsEnum.VP8) }, MediaStreamStatusEnum.Inactive);
//peerConnection.addTrack(videoTrack);
var answerResult = await userAgent.Answer(uas, peerConnection);
};
Console.Write("press any key to exit...");
Console.Read();
sipTransport.Shutdown();
}