/*
* /// <summary>
* /// Transfers call to specified recipient.
* /// </summary>
* /// <param name="to">Address where to transfer call.</param>
* public void CallTransfer(string to)
* {
* throw new NotImplementedException();
* }*/
#endregion
#region method CopyMessage
/// <summary>
/// Copies header fileds from 1 message to antother.
/// </summary>
/// <param name="source">Source message.</param>
/// <param name="destination">Destination message.</param>
/// <param name="exceptHeaders">Header fields not to copy.</param>
private void CopyMessage(SIP_Message source, SIP_Message destination, string[] exceptHeaders)
{
foreach (SIP_HeaderField headerField in source.Header)
{
bool copy = true;
foreach (string h in exceptHeaders)
{
if (h.ToLower() == headerField.Name.ToLower())
{
copy = false;
break;
}
}
if (copy)
{
destination.Header.Add(headerField.Name, headerField.Value);
}
}
destination.Data = source.Data;
}
/*
/// <summary>
/// Transfers call to specified recipient.
/// </summary>
/// <param name="to">Address where to transfer call.</param>
public void CallTransfer(string to)
{
throw new NotImplementedException();
}*/
/// <summary>
/// Copies header fileds from 1 message to antother.
/// </summary>
/// <param name="source">Source message.</param>
/// <param name="destination">Destination message.</param>
/// <param name="exceptHeaders">Header fields not to copy.</param>
private void CopyMessage(SIP_Message source, SIP_Message destination, string[] exceptHeaders)
{
foreach (SIP_HeaderField headerField in source.Header)
{
bool copy = true;
foreach (string h in exceptHeaders)
{
if (h.ToLower() == headerField.Name.ToLower())
{
copy = false;
break;
}
}
if (copy)
{
destination.Header.Add(headerField.Name, headerField.Value);
}
}
destination.Data = source.Data;
}
private static void device_OnPacketArrival(object sender, CaptureEventArgs e)
{
var time = e.Packet.Timeval.Date;
var len = e.Packet.Data.Length;
var packet = PacketDotNet.Packet.ParsePacket(e.Packet.LinkLayerType, e.Packet.Data);
var udpPacket = PacketDotNet.UdpPacket.GetEncapsulated(packet);
if (udpPacket != null)
{
try
{
// signalling packet
SIP_Message msg = ParseSIPMessage(udpPacket.PayloadData);
if (msg != null && msg.CallID != null)
{
SDP_Message sdp = null;
try
{
sdp = SDP_Message.Parse(System.Text.ASCIIEncoding.Default.GetString(msg.Data));
}
catch { }
if (msg is SIP_Request && msg.CallID != null)
{
SIP_Request r = (SIP_Request)msg;
if (!Call.Calls.ContainsKey(r.CallID))
{
if (r.RequestLine.Method == "INVITE")
{
Call.Calls.Add(r.CallID, new Call(r.CallID));
Call.Calls[r.CallID].CallerIP = ((IpPacket)udpPacket.ParentPacket).SourceAddress;
Call.Calls[r.CallID].CalleeIP = ((IpPacket)udpPacket.ParentPacket).DestinationAddress;
}
else
{
return; // Ignore this conversation
}
}
// if this is an invite, do we have an audio rtp port defined?
if (r.RequestLine.Method == "INVITE")
{
if (sdp != null)
{
foreach (var a in sdp.MediaDescriptions)
{
Console.Out.WriteLine(r.CallID + " - Got RTP Media Port: " + ((IpPacket)udpPacket.ParentPacket).SourceAddress + ":" + a.Port.ToString());
if (Call.Calls[r.CallID].CallerIP.ToString() == ((IpPacket)udpPacket.ParentPacket).SourceAddress.ToString())
{
Call.Calls[r.CallID].CallerRTPPort = a.Port;
}
else
{
Call.Calls[r.CallID].CalleeRTPPort = a.Port;
}
}
}
}
if (r.RequestLine.Method == "BYE")
{
if (Call.Calls.ContainsKey(r.CallID))
{
// Log bye was recevied
Call.Calls[r.CallID].SeenBYE = true;
// Now indicate who hung up
Call.Calls[r.CallID].WhoHungUp = ((IpPacket)udpPacket.ParentPacket).SourceAddress == Call.Calls[r.CallID].CallerIP ?
Call.CallDirection.Caller : Call.CallDirection.Callee;
}
else
{
Console.WriteLine("Unknown CallID: " + r.CallID);
}
}
}
else if (msg is SIP_Response && msg.CallID != null)
{
SIP_Response r = (SIP_Response)msg;
if (sdp != null)
{
foreach (var a in sdp.MediaDescriptions)
{
Console.Out.WriteLine(r.CallID + " - Got RTP Media Port: " + ((IpPacket)udpPacket.ParentPacket).SourceAddress + ":" + a.Port.ToString());
if (Call.Calls[r.CallID].CallerIP.ToString() == ((IpPacket)udpPacket.ParentPacket).SourceAddress.ToString())
{
Call.Calls[r.CallID].CallerRTPPort = a.Port;
}
else
{
Call.Calls[r.CallID].CalleeRTPPort = a.Port;
}
}
}
if (Call.Calls.ContainsKey(r.CallID))
{
if (r.StatusCodeType == SIP_StatusCodeType.Success && Call.Calls[r.CallID].SeenBYE)
{
Call.Calls[r.CallID].Confirmed = true;
}
}
}
// Add packet to history
if (Call.Calls.ContainsKey(msg.CallID))
{
Call.Calls[msg.CallID].WritePacket(e.Packet, Call.PacketType.SIPDialog);
// Check to see is this call has been terminated
if (Call.Calls[msg.CallID].Confirmed)
{
// Close off the call now last data has been written
Console.WriteLine("Call Ended: " + msg.CallID);
// Close off the call
Call.Calls[msg.CallID].CloseCall();
// Remove the call from the in-memory list
Call.Calls.Remove(msg.CallID);
}
}
}
else
{
Call c = Call.GetCallByRTPPort(udpPacket.SourcePort);
if (c != null)
{
c.WritePacket(e.Packet, Call.PacketType.RTP);
}
}
}
catch (Exception ex) {
Console.WriteLine(ex.ToString());
}
}
}
public void Handler(Packet packet)
{
var udpPacket = UdpPacket.GetEncapsulated(packet);
// if it's not udp , udpPacket will be null and we don't handle it.
if (udpPacket != null)
{
try
{
// signalling packet
SIP_Message msg = ParseSIPMessage(udpPacket.PayloadData);
if (msg != null && msg.CallID != null)
{
SDP_Message sdp = null;
Console.WriteLine("SIP capture");
try
{
sdp = SDP_Message.Parse(System.Text.Encoding.Default.GetString(msg.Data));
}
catch { }
if (msg is SIP_Request && msg.CallID != null)
{
SIP_Request r = (SIP_Request)msg;
//already containsKey
if (!Call.SIPSessions.ContainsKey(r.CallID))
{
if (r.RequestLine.Method == "INVITE")
{
Call.SIPSessions.Add(r.CallID, new Call(r.CallID));
Call.SIPSessions[r.CallID].CallerIP = ((IpPacket)udpPacket.ParentPacket).SourceAddress;
Call.SIPSessions[r.CallID].CalleeIP = ((IpPacket)udpPacket.ParentPacket).DestinationAddress;
}
else
{
return; // Ignore this conversation
}
}
// if this is an invite, do we have an audio rtp port defined?
if (r.RequestLine.Method == "INVITE")
{
if (sdp != null)
{
foreach (var a in sdp.MediaDescriptions)
{
Console.Out.WriteLine(r.CallID + " - Got RTP Media Port: " + ((IpPacket)udpPacket.ParentPacket).SourceAddress + ":" + a.Port.ToString());
if (Call.SIPSessions[r.CallID].CallerIP.ToString() == ((IpPacket)udpPacket.ParentPacket).SourceAddress.ToString())
{
Call.SIPSessions[r.CallID].CallerRTPPort = a.Port;
}
else
{
Call.SIPSessions[r.CallID].CalleeRTPPort = a.Port;
}
a.MediaFormats.GetType();
break; // First description is about audio . Second is about viedo and we don't need it, so break.
}
}
}
if (r.RequestLine.Method == "BYE")
{
if (Call.SIPSessions.ContainsKey(r.CallID))
{
// Log bye was recevied
Call.SIPSessions[r.CallID].SeenBYE = true;
// Now indicate who hung up
Call.SIPSessions[r.CallID].WhoHungUp = ((IpPacket)udpPacket.ParentPacket).SourceAddress == Call.SIPSessions[r.CallID].CallerIP ?
Call.CallDirection.Caller : Call.CallDirection.Callee;
}
else
{
Console.WriteLine("Unknown CallID: " + r.CallID);
}
}
}// if (msg is SIP_Request && msg.CallID != null)
else if (msg is SIP_Response && msg.CallID != null)
{
SIP_Response r = (SIP_Response)msg;
if (r.StatusCode != 183 && r.StatusCode != 100 && r.StatusCode != 200)
{
Call.SIPSessions[r.CallID].isEnd = true;
}
if (sdp != null)
{
foreach (var a in sdp.MediaDescriptions)
{
Console.Out.WriteLine(r.CallID + " - Got RTP Media Port: " + ((IpPacket)udpPacket.ParentPacket).SourceAddress + ":" + a.Port.ToString());
if (Call.SIPSessions[r.CallID].CallerIP.ToString() == ((IpPacket)udpPacket.ParentPacket).SourceAddress.ToString())
{
Call.SIPSessions[r.CallID].CallerRTPPort = a.Port;
}
else
{
Call.SIPSessions[r.CallID].CalleeRTPPort = a.Port;
}
break; // First description is about audio . Second is about viedo and we don't need it, so break.
}
}
if (Call.SIPSessions.ContainsKey(r.CallID))
{
if (r.StatusCodeType == SIP_StatusCodeType.Success && Call.SIPSessions[r.CallID].SeenBYE)
{
Call.SIPSessions[r.CallID].Confirmed = true;
Call.SIPSessions[r.CallID].isEnd = true;
}
}
}
// Add packet to history
if (Call.SIPSessions.ContainsKey(msg.CallID))
{
Call.SIPSessions[msg.CallID].WritePacket(packet, Call.PacketType.SIPDialog);
// Check to see is this call has been terminated
if (Call.SIPSessions[msg.CallID].Confirmed)
{
// Close off the call now last data has been written
Console.WriteLine("Call Ended: " + msg.CallID);
// Close off the call
Call.SIPSessions[msg.CallID].CloseCall();
StringBuilder file = new StringBuilder(Directory.GetCurrentDirectory() + "//" + Call.SIPSessions[msg.CallID].SIPPacketFilePathAndName);
StringBuilder StoragePath = new StringBuilder(Directory.GetCurrentDirectory() + "//" + Call.SIPSessions[msg.CallID].SIPPacketFilePath);
pacp_to_wav(file, StoragePath);
}
if (Call.SIPSessions[msg.CallID].isEnd == true)
{
Call.SIPSessions.Remove(msg.CallID);
}
}
}
else
{
Call c = Call.GetCallByRTPPort(udpPacket.SourcePort);
if (c != null)
{
c.WritePacket(packet, Call.PacketType.RTP);
}
}
}
catch (Exception ex)
{
Console.WriteLine(ex.ToString());
}
}
}