/// <summary>
/// Default incoming call constructor.
/// </summary>
/// <param name="ua">Owner UA.</param>
/// <param name="invite">INVITE server transaction.</param>
/// <exception cref="ArgumentNullException">Is riased when <b>ua</b> or <b>invite</b> is null reference.</exception>
internal SIP_UA_Call(SIP_UA ua, SIP_ServerTransaction invite)
{
if (ua == null)
{
throw new ArgumentNullException("ua");
}
if (invite == null)
{
throw new ArgumentNullException("invite");
}
m_pUA = ua;
m_pInitialInviteTransaction = invite;
m_pLocalUri = invite.Request.To.Address.Uri;
m_pRemoteUri = invite.Request.From.Address.Uri;
m_pInitialInviteTransaction.Canceled += new EventHandler(delegate(object sender, EventArgs e){
// If transaction canceled, terminate call.
SetState(SIP_UA_CallState.Terminated);
});
// Parse SDP if INVITE contains SDP.
// RFC 3261 13.2.1. INVITE may be offerless, we must thne send offer and remote party sends sdp in ACK.
if (invite.Request.ContentType != null && invite.Request.ContentType.ToLower().IndexOf("application/sdp") > -1)
{
m_pRemoteSDP = SDP_Message.Parse(Encoding.UTF8.GetString(invite.Request.Data));
}
m_pTags = new Dictionary <string, object>();
m_State = SIP_UA_CallState.WaitingToAccept;
}
private void toResponse(SIP_RequestReceivedEventArgs e)
{
SIP_Uri uri = e.Request.RequestLine.Uri as SIP_Uri;
try
{
_sdp = SDP_Message.Parse(MyEncoder.Encoder.GetString(e.Request.Data));
RemoteIP = _sdp.Origin.UnicastAddress;
if (_sdp.MediaDescriptions.Count == 0)
{
e.ServerTransaction.SendResponse(_sipServer.Stack.CreateResponse(SIP_ResponseCodes.x400_Bad_Request, e.Request));
return;
}
else
{
RemotePort = _sdp.MediaDescriptions[0].Port;
}
}
catch (Exception)
{
//解析SDP失败。
e.ServerTransaction.SendResponse(_sipServer.Stack.CreateResponse(SIP_ResponseCodes.x400_Bad_Request, e.Request));
return;
}
//send 100 Trying;
e.ServerTransaction.SendResponse(_sipServer.Stack.CreateResponse(SIP_ResponseCodes.x100_Trying, e.Request));
_videoId = _sipServer.DeviceManager.GetVideoId(uri.User);
if (_videoId != null)
{
RTPServer rtp = _sipServer.RTPManager.GetOrAddServer(_videoId);
SDP_Message respSDP = new SDP_Message();
respSDP.Version = "0";
respSDP.Origin = new SDP_Origin(uri.User, 0, 0, "IN", "IPV4", rtp.LocalIP);
respSDP.SessionName = "Play";
respSDP.Connection = new SDP_Connection("IN", "IPV4", rtp.LocalIP);
respSDP.SSRC = SDP_Utils.SSRC2String(SDP_Utils.GenSSRC(uri.User, true)); //根据国标补充协议标准生成SSRC。
respSDP.Times.Add(new SDP_Time(0, 0));
respSDP.MediaDescriptions.Add(new SDP_MediaDescription("video", rtp.Port, 2, "RTP/AVP", new string[] { "96", "97", "98" }));
respSDP.Attributes.Add(new SDP_Attribute("sendonly", ""));
respSDP.Attributes.Add(new SDP_Attribute("rtpmap", "96 PS/90000"));
respSDP.Attributes.Add(new SDP_Attribute("rtpmap", "97 MPEG4/90000"));
respSDP.Attributes.Add(new SDP_Attribute("rtpmap", "98 H264/90000"));
SIP_Response resp = _sipServer.Stack.CreateResponse(SIP_ResponseCodes.x200_Ok, e.Request);
resp.Data = respSDP.ToByte();
e.ServerTransaction.SendResponse(resp);
}
else
{
//没有找到视频源。
e.ServerTransaction.SendResponse(_sipServer.Stack.CreateResponse(SIP_ResponseCodes.x404_Not_Found, e.Request));
}
}
/// <summary>
/// This method is called when initial INVITE sender got response.
/// </summary>
/// <param name="sender">Sender.</param>
/// <param name="e">Event data.</param>
private void m_pInitialInviteSender_ResponseReceived(object sender, SIP_ResponseReceivedEventArgs e)
{
try{
lock (m_pLock){
// If remote party provided SDP, parse it.
if (e.Response.ContentType != null && e.Response.ContentType.ToLower().IndexOf("application/sdp") > -1)
{
m_pRemoteSDP = SDP_Message.Parse(Encoding.UTF8.GetString(e.Response.Data));
// TODO: If parsing failed, end call.
}
if (e.Response.StatusCodeType == SIP_StatusCodeType.Provisional)
{
if (e.Response.StatusCode == 180)
{
SetState(SIP_UA_CallState.Ringing);
}
else if (e.Response.StatusCode == 182)
{
SetState(SIP_UA_CallState.Queued);
}
// We don't care other status responses.
/* RFC 3261 13.2.2.1.
* Zero, one or multiple provisional responses may arrive before one or
* more final responses are received. Provisional responses for an
* INVITE request can create "early dialogs". If a provisional response
* has a tag in the To field, and if the dialog ID of the response does
* not match an existing dialog, one is constructed using the procedures
* defined in Section 12.1.2.
*/
if (e.Response.StatusCode > 100 && e.Response.To.Tag != null)
{
m_pEarlyDialogs.Add((SIP_Dialog_Invite)m_pUA.Stack.TransactionLayer.GetOrCreateDialog(e.ClientTransaction, e.Response));
}
}
else if (e.Response.StatusCodeType == SIP_StatusCodeType.Success)
{
m_StartTime = DateTime.Now;
SetState(SIP_UA_CallState.Active);
m_pDialog = m_pUA.Stack.TransactionLayer.GetOrCreateDialog(e.ClientTransaction, e.Response);
m_pDialog.StateChanged += new EventHandler(m_pDialog_StateChanged);
/* Exit all all other dialogs created by this call (due to forking).
* That is not defined in RFC but, since UAC can send BYE to early and confirmed dialogs,
* because of this all 100% valid.
*/
foreach (SIP_Dialog_Invite dialog in m_pEarlyDialogs.ToArray())
{
if (!m_pDialog.Equals(dialog))
{
dialog.Terminate("Another forking leg accepted.", true);
}
}
}
else
{
/* RFC 3261 13.2.2.3.
* All early dialogs are considered terminated upon reception of the non-2xx final response.
*/
foreach (SIP_Dialog_Invite dialog in m_pEarlyDialogs.ToArray())
{
dialog.Terminate("All early dialogs are considered terminated upon reception of the non-2xx final response. (RFC 3261 13.2.2.3)", false);
}
m_pEarlyDialogs.Clear();
Error();
SetState(SIP_UA_CallState.Terminated);
}
}
}
catch (Exception x) {
m_pUA.Stack.OnError(x);
}
}
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());
}
}
}
