private void m_pToolbar_ItemClicked(object sender, ToolStripItemClickedEventArgs e)
{
if (e.ClickedItem.Tag == null)
{
return;
}
if (e.ClickedItem.Tag.ToString() == "delete")
{
SipCall sipCall = (SipCall)this.m_pCalls.SelectedItems[0].Tag;
if (MessageBox.Show(this, string.Concat(new string[]
{
"Are you sure you want to terminate call '",
sipCall.Caller,
"->",
sipCall.Callee,
"' ?"
}), "Remove Registration", MessageBoxButtons.YesNo, MessageBoxIcon.Question) == DialogResult.Yes)
{
sipCall.Terminate();
this.m_pCalls.SelectedItems[0].Remove();
return;
}
}
else if (e.ClickedItem.Tag.ToString() == "refresh")
{
this.LoadData();
}
}
static void test1()
{
IvrWORX x = new IvrWORX();
x.Init("dotnet.json");
StreamerSession streamer = new StreamerSession(x);
streamer.Allocate(null,
StreamerSession.RcvDeviceType.RCV_DEVICE_NONE,
StreamerSession.SndDeviceType.SND_DEVICE_TYPE_FILE);
SipCall sipCall = new SipCall(x);
Credentials c = new Credentials();
c.User = "******";
c.Realm = "example.com";
c.Password = "******";
LinkedList <string> contactList =
new LinkedList <string>();
contactList.AddLast("sip:1234" + "@" + GetLocalIP());
// ApiErrorCode res = sipCall.StartRegistration(
// contactList,
// "sip:[email protected]",
// c,
// DEFAULT_TIMEOUT);
//
// if (res != ApiErrorCode.API_SUCCESS)
// throw new Exception("Registration failed err:" + res);
ApiErrorCode res = sipCall.MakeCall("sip:[email protected]", streamer.LocalOffer(), c, null, DEFAULT_TIMEOUT);
streamer.ModifyConnection(sipCall.RemoteOffer());
streamer.PlayFile(@"c:\SOUNDS\greeting.wav", true, false);
sipCall.HangupCall();
Console.WriteLine("MakeCall res=" + res);
}
public SipCall NewSipCall(string to, string from, string fromName = null, string codecs = null, JArray headers = null, int timeout = 30, int?maxDuration = null, bool?webRTCMedia = null)
{
SipCall call = new SipCall(this, Guid.NewGuid().ToString())
{
To = to,
From = from,
FromName = fromName,
Headers = headers,
Codecs = codecs,
Timeout = timeout,
MaxDuration = maxDuration,
WebRTCMedia = webRTCMedia
};
mCalls.TryAdd(call.TemporaryID, call);
OnCallCreated?.Invoke(this, call);
return(call);
}
private void OnCallingEvent_Receive(Client client, BroadcastParams broadcastParams, CallingEventParams callEventParams)
{
CallingEventParams.ReceiveParams receiveParams = null;
try { receiveParams = callEventParams.ParametersAs <CallingEventParams.ReceiveParams>(); }
catch (Exception exc)
{
Log(LogLevel.Warning, exc, "Failed to parse ReceiveParams");
return;
}
// @note A received call should only ever receive one receive event regardless of the state of the call, but we act as though
// we could receive multiple just for sanity here, but the callbacks will still only be called when first created, which makes
// this effectively a no-op on additional receive events
Call call = null;
Call tmp = null;
switch (receiveParams.Device.Type)
{
case CallDevice.DeviceType.phone:
{
CallDevice.PhoneParams phoneParams = null;
try { phoneParams = receiveParams.Device.ParametersAs <CallDevice.PhoneParams>(); }
catch (Exception exc)
{
Log(LogLevel.Warning, exc, "Failed to parse PhoneParams");
return;
}
// If the call already exists under the real call id simply obtain the call, otherwise create a new call
call = mCalls.GetOrAdd(receiveParams.CallID, k => (tmp = new PhoneCall(this, receiveParams.NodeID, receiveParams.CallID)
{
To = phoneParams.ToNumber,
From = phoneParams.FromNumber,
Timeout = phoneParams.Timeout,
}));
break;
}
case CallDevice.DeviceType.sip:
{
CallDevice.SipParams sipParams = null;
try { sipParams = receiveParams.Device.ParametersAs <CallDevice.SipParams>(); }
catch (Exception exc)
{
Log(LogLevel.Warning, exc, "Failed to parse SipParams");
return;
}
// If the call already exists under the real call id simply obtain the call, otherwise create a new call
call = mCalls.GetOrAdd(receiveParams.CallID, k => (tmp = new SipCall(this, receiveParams.NodeID, receiveParams.CallID)
{
To = sipParams.To,
From = sipParams.From,
Headers = sipParams.Headers,
}));
break;
}
// @TODO: webrtc
default:
Log(LogLevel.Warning, string.Format("Unknown device type: {0}", receiveParams.Device.Type));
return;
}
if (tmp == call)
{
call.Context = receiveParams.Context;
OnCallCreated?.Invoke(this, call);
OnCallReceived?.Invoke(this, call, receiveParams);
}
call.ReceiveHandler(callEventParams, receiveParams);
}
private void OnCallingEvent_State(Client client, BroadcastParams broadcastParams, CallingEventParams callEventParams)
{
CallingEventParams.StateParams stateParams = null;
try { stateParams = callEventParams.ParametersAs <CallingEventParams.StateParams>(); }
catch (Exception exc)
{
Log(LogLevel.Warning, exc, "Failed to parse StateParams");
return;
}
Call call = null;
Call tmp = null;
switch (stateParams.Device.Type)
{
case CallDevice.DeviceType.phone:
{
CallDevice.PhoneParams phoneParams = null;
try { phoneParams = stateParams.Device.ParametersAs <CallDevice.PhoneParams>(); }
catch (Exception exc)
{
Log(LogLevel.Warning, exc, "Failed to parse PhoneParams");
return;
}
// If the call already exists under the real call id simply obtain the call, however if the call was found under
// a temporary call id then readd it here under the real call id, otherwise create a new call
call = mCalls.GetOrAdd(stateParams.CallID, k => call ?? (tmp = new PhoneCall(this, stateParams.NodeID, stateParams.CallID)
{
To = phoneParams.ToNumber,
From = phoneParams.FromNumber,
Timeout = phoneParams.Timeout,
// Capture the state, it may not always be created the first time we see the call
State = stateParams.CallState,
}));
break;
}
case CallDevice.DeviceType.sip:
{
CallDevice.SipParams sipParams = null;
try { sipParams = stateParams.Device.ParametersAs <CallDevice.SipParams>(); }
catch (Exception exc)
{
Log(LogLevel.Warning, exc, "Failed to parse SipParams");
return;
}
// If the call already exists under the real call id simply obtain the call, however if the call was found under
// a temporary call id then readd it here under the real call id, otherwise create a new call
call = mCalls.GetOrAdd(stateParams.CallID, k => call ?? (tmp = new SipCall(this, stateParams.NodeID, stateParams.CallID)
{
To = sipParams.To,
From = sipParams.From,
FromName = sipParams.FromName,
Headers = sipParams.Headers,
// Capture the state, it may not always be created the first time we see the call
State = stateParams.CallState,
}));
break;
}
// @TODO: webrtc
default:
Log(LogLevel.Warning, string.Format("Unknown device type: {0}", stateParams.Device.Type));
return;
}
if (tmp == call)
{
OnCallCreated?.Invoke(this, call);
}
call.StateChangeHandler(callEventParams, stateParams);
}
static void test2()
{
IvrWORX x = new IvrWORX();
x.Init("dotnet.json");
RtpProxySession streamerRtpSession = new RtpProxySession(x);
AbstractOffer dummyOffer = new AbstractOffer();
dummyOffer.Type = "application/sdp";
dummyOffer.Body = @"v=0" + "\n"
+ "o=alice 2890844526 2890844526 IN IP4 0.0.0.0" + "\n"
+ "s=" + "\n"
+ "c=IN IP4 0.0.0.0" + "\n"
+ "t=0 0" + "\n"
+ "m=audio 0 RTP/AVP 0" + "\n"
+ "a=rtpmap:0 PCMU/8000" + "\n\n";;
ApiErrorCode res = streamerRtpSession.Allocate(dummyOffer);
if (res != ApiErrorCode.API_SUCCESS)
{
throw new Exception("RtpProxySession(1) Allocated failed err:" + res);
}
StreamerSession streamer = new StreamerSession(x);
streamer.Allocate(streamerRtpSession.LocalOffer(),
StreamerSession.RcvDeviceType.RCV_DEVICE_NONE,
StreamerSession.SndDeviceType.SND_DEVICE_TYPE_FILE);
if (res != ApiErrorCode.API_SUCCESS)
{
throw new Exception("StreamerSession Allocated failed err:" + res);
}
streamerRtpSession.Modify(streamer.LocalOffer());
RtpProxySession sipRtpSession = new RtpProxySession(x);
sipRtpSession.Allocate(dummyOffer);
if (res != ApiErrorCode.API_SUCCESS)
{
throw new Exception("RtpProxySession(2) Allocated failed err:" + res);
}
SipCall sipCall = new SipCall(x);
res =
sipCall.MakeCall("sip:[email protected]", sipRtpSession.LocalOffer(), null, null, DEFAULT_TIMEOUT);
if (res != ApiErrorCode.API_SUCCESS)
{
throw new Exception("MakeCall failed err:" + res);
}
sipRtpSession.Modify(sipCall.RemoteOffer());
sipRtpSession.Bridge(streamerRtpSession, true);
if (res != ApiErrorCode.API_SUCCESS)
{
throw new Exception("Bridge failed err:" + res);
}
streamer.PlayFile(@"c:\SOUNDS\greeting.wav", true, false);
sipCall.Dispose();
Console.WriteLine("MakeCall res=" + res);
}
static void test6()
{
IvrWORX x = new IvrWORX();
x.Init("dotnet.json");
RtpProxySession rtpProxy =
new RtpProxySession(x);
//
// Allocate RTP endpoint for RTSP session.
//
RtpProxySession call1RtpSession = new RtpProxySession(x);
AbstractOffer dummyOffer = new AbstractOffer();
dummyOffer.Type = "application/sdp";
dummyOffer.Body = @"v=0
o=alice 2890844526 2890844526 IN IP4 0.0.0.0
s=
c=IN IP4 0.0.0.0
t=0 0
m=audio 0 RTP/AVP 8 101
a=rtpmap:0 PCMA/8000
a=rtpmap:101 telephone-event/8000
";
ApiErrorCode res = call1RtpSession.Allocate(dummyOffer);
if (res != ApiErrorCode.API_SUCCESS)
{
throw new Exception("RtpProxySession(1) Allocated failed err:" + res);
}
SipCall caller = new SipCall(x);
res = caller.MakeCall("sip:[email protected]",
call1RtpSession.LocalOffer(),
null,
null,
DEFAULT_TIMEOUT);
if (res != ApiErrorCode.API_SUCCESS)
{
throw new Exception("MakeCall err:" + res);
}
bool shutdown = false;
List <IActiveObject> actors =
new List <IActiveObject>();
actors.Add(caller);
actors.Add(call1RtpSession);
while (!shutdown)
{
Selector s = new Selector();
int index = 0;
int event_id = 0;
res = s.Select(
actors,
DEFAULT_TIMEOUT,
ref index,
ref event_id);
String dtmfs = call1RtpSession.DtmfBuffer();
Console.Out.WriteLine("err=" + res +
", index=" + index +
", event=" + event_id +
", dtmfs=" + dtmfs);
if (dtmfs.Contains("#") || index == 0)
{
shutdown = true;
}
}
caller.HangupCall();
}
static void test5()
{
IvrWORX x = new IvrWORX();
x.Init("dotnet.json");
SipCall caller = new SipCall(x);
ApiErrorCode res = caller.MakeCall("sip:[email protected]", null, null, null, DEFAULT_TIMEOUT);
if (res != ApiErrorCode.API_SUCCESS)
{
throw new Exception("MakeCall err:" + res);
}
MrcpSession recog =
new MrcpSession(x);;
recog.Allocate(MrcpResource.RECOGNIZER, caller.RemoteOffer(), DEFAULT_TIMEOUT);
if (res != ApiErrorCode.API_SUCCESS)
{
throw new Exception("Allocate err:" + res);
}
caller.Answer(recog.RemoteOffer(MrcpResource.RECOGNIZER), null, DEFAULT_TIMEOUT);
if (res != ApiErrorCode.API_SUCCESS)
{
throw new Exception("Allocate err:" + res);
}
MrcpSession synth = new MrcpSession(x);
synth.Allocate(MrcpResource.SYNTHESIZER, caller.RemoteOffer(), DEFAULT_TIMEOUT);
if (res != ApiErrorCode.API_SUCCESS)
{
throw new Exception("Allocate(SYNTHESIZER) err:" + res);
}
Thread.Sleep(1000);
synth.SimpleSpeak(null, "Please choose red pill, or blue pill", true);
if (res != ApiErrorCode.API_SUCCESS)
{
throw new Exception("Speak err:" + res);
}
String pillgrammar = @"<?xml version=""1.0"" encoding=""UTF-8""?>
<!-- the default grammar language is US English -->
<grammar xmlns=""http://www.w3.org/2001/06/grammar""
xmlns:xsi=""http://www.w3.org/2001/XMLSchema-instance""
xml:lang=""en-US"" version=""1.0"" root=""pill"">
<rule id=""pill"">
<one-of>
<item>red</item>
<item>blue</item>
</one-of>
</rule>
</grammar>";
Dictionary <String, Object> p =
new Dictionary <String, Object>();
p["cancel_if_queue"] = true;
p["content_id"] = "<*****@*****.**>";
p["content_type"] = "application/srgs+xml";
p["no_input_timeout"] = 20000;
p["confidence_threshhold"] = 0.9;
p["start_input_timers"] = true;
String answer;
res = recog.Recognize(
p,
pillgrammar,
DEFAULT_TIMEOUT * 10000,
true,
out answer);
if (res != ApiErrorCode.API_SUCCESS)
{
throw new Exception("Speak err:" + res);
}
Console.Out.WriteLine(answer);
caller.WaitTillHangup();
}
static void test4()
{
IvrWORX x = new IvrWORX();
x.Init("dotnet.json");
//
// Allocate RTP endpoint for RTSP session.
//
RtpProxySession call1RtpSession = new RtpProxySession(x);
AbstractOffer dummyOffer = new AbstractOffer();
dummyOffer.Type = "application/sdp";
dummyOffer.Body = @"v=0" + "\n"
+ "o=alice 2890844526 2890844526 IN IP4 0.0.0.0" + "\n"
+ "s=" + "\n"
+ "c=IN IP4 0.0.0.0" + "\n"
+ "t=0 0" + "\n"
+ "m=audio 0 RTP/AVP 8" + "\n"
+ "a=rtpmap:0 PCMA/8000" + "\n\n";;
ApiErrorCode res = call1RtpSession.Allocate(dummyOffer);
if (res != ApiErrorCode.API_SUCCESS)
{
throw new Exception("RtpProxySession(1) Allocated failed err:" + res);
}
//
// Allocate sip RTP session
//
RtpProxySession call2RtpSession = new RtpProxySession(x);
res = call2RtpSession.Allocate(dummyOffer);
if (res != ApiErrorCode.API_SUCCESS)
{
throw new Exception("RtpProxySession(1) Allocated failed err:" + res);
}
//
// Make call1
//
SipCall sipCall1 = new SipCall(x);
res = sipCall1.MakeCall("sip:[email protected]", call1RtpSession.LocalOffer(), null, null, DEFAULT_TIMEOUT);
if (res != ApiErrorCode.API_SUCCESS)
{
throw new Exception("MakeCall failed err:" + res);
}
call1RtpSession.Modify(sipCall1.RemoteOffer());
if (res != ApiErrorCode.API_SUCCESS)
{
throw new Exception("Modify failed err:" + res);
}
//
// Make call2
//
SipCall sipCall2 = new SipCall(x);
res = sipCall2.MakeCall("sip:[email protected]", call2RtpSession.LocalOffer(), null, null, DEFAULT_TIMEOUT);
if (res != ApiErrorCode.API_SUCCESS)
{
throw new Exception("MakeCall failed err:" + res);
}
call2RtpSession.Modify(sipCall2.RemoteOffer());
call1RtpSession.Bridge(call2RtpSession, true);
if (res != ApiErrorCode.API_SUCCESS)
{
throw new Exception("Bridge failed err:" + res);
}
Thread.Sleep(30000);
sipCall1.Dispose();
sipCall2.Dispose();
Console.WriteLine("MakeCall res=" + res);
}
static void test3()
{
IvrWORX x = new IvrWORX();
x.Init("dotnet.json");
//
// Allocate RTP endpoint for RTSP session.
//
RtpProxySession rtspRtpSession = new RtpProxySession(x);
AbstractOffer dummyOffer = new AbstractOffer();
dummyOffer.Type = "application/sdp";
dummyOffer.Body = @"v=0" + "\n"
+ "o=alice 2890844526 2890844526 IN IP4 0.0.0.0" + "\n"
+ "s=" + "\n"
+ "c=IN IP4 0.0.0.0" + "\n"
+ "t=0 0" + "\n"
+ "m=audio 0 RTP/AVP 8" + "\n"
+ "a=rtpmap:0 PCMA/8000" + "\n\n";;
ApiErrorCode res = rtspRtpSession.Allocate(dummyOffer);
if (res != ApiErrorCode.API_SUCCESS)
{
throw new Exception("RtpProxySession(1) Allocated failed err:" + res);
}
//
// Set up RTSP session and update its RTP endpoint.
//
RtspSession rtsp = new RtspSession(x);
res = rtsp.Setup("rtsp://10.116.100.78/IvrScript.wav", rtspRtpSession.LocalOffer());
if (res != ApiErrorCode.API_SUCCESS)
{
throw new Exception("RtspSession Setup failed err:" + res);
}
res = rtspRtpSession.Modify(rtsp.RemoteOffer());
if (res != ApiErrorCode.API_SUCCESS)
{
throw new Exception("RtspSession Modify failed err:" + res);
}
//
// Allocate sip RTP session
//
RtpProxySession sipRtpSession = new RtpProxySession(x);
res = sipRtpSession.Allocate(dummyOffer);
if (res != ApiErrorCode.API_SUCCESS)
{
throw new Exception("RtpProxySession(1) Allocated failed err:" + res);
}
//
// Make call
//
SipCall sipCall = new SipCall(x);
res = sipCall.MakeCall("sip:[email protected]", sipRtpSession.LocalOffer(), null, null, DEFAULT_TIMEOUT);
if (res != ApiErrorCode.API_SUCCESS)
{
throw new Exception("MakeCall failed err:" + res);
}
sipRtpSession.Modify(sipCall.RemoteOffer());
if (res != ApiErrorCode.API_SUCCESS)
{
throw new Exception("Modify failed err:" + res);
}
rtspRtpSession.Bridge(sipRtpSession, false);
if (res != ApiErrorCode.API_SUCCESS)
{
throw new Exception("Bridge failed err:" + res);
}
res = rtsp.Play(0.0, 0.0, 1);
if (res != ApiErrorCode.API_SUCCESS)
{
throw new Exception("RtpProxySession(1) Allocated failed err:" + res);
}
Thread.Sleep(10000);
sipCall.Dispose();
Console.WriteLine("MakeCall res=" + res);
}
