sipdialogconfvoip.cxx

MiniSip Client with DomainKeys Authentication, Sip, Audio communications, Echo Cancel

/*
 Copyright (C) 2004-2006 the Minisip Team
 
 This library is free software; you can redistribute it and/or
 modify it under the terms of the GNU Lesser General Public
 License as published by the Free Software Foundation; either
 version 2.1 of the License, or (at your option) any later version.
 
 This library is distributed in the hope that it will be useful,
 but WITHOUT ANY WARRANTY; without even the implied warranty of
 MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
 Lesser General Public License for more details.
 
 You should have received a copy of the GNU Lesser General Public
 License along with this library; if not, write to the Free Software
 Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307 USA
 */

/* Copyright (C) 2004 
 *
 * Authors: Erik Eliasson <[e-mail]>
 *	Johan Bilien <[e-mail]>
 *	Joachim Orrblad <joachim[at]orrblad.com>
 */	    

/* Name
 * 	SipDialogConfVoip.cxx
 * Authors
 * 	Erik Eliasson, [e-mail]
 	Bilge Cetin <bilge[at]kth.se>
	Max Loubser <loubser[at]kth.se>
*/

#include<config.h>

#include<libminisip/sip/SipDialogConfVoip.h>

#include<libmutil/massert.h>

//#include<libmsip/SipDialogContainer.h>
#include<libmsip/SipResponse.h>
#include<libmsip/SipTransactionInviteClientUA.h>
#include<libmsip/SipTransactionInviteServerUA.h>
#include<libmsip/SipTransactionNonInviteClient.h>
#include<libmsip/SipTransactionNonInviteServer.h>
#include<libmsip/SipTransactionUtils.h>
#include<libmsip/SipDialog.h>
#include<libmsip/SipCommandString.h>
#include<libmsip/SipHeaderTo.h>
#include<libmsip/SipHeaderAcceptContact.h>
#include<libmsip/SipHeaderFrom.h>
#include<libmsip/SipHeaderWarning.h>
#include<libmsip/SipMIMEContent.h>
#include<libmsip/SipMessageContent.h>
#include<libminisip/sip/DefaultDialogHandler.h>
#include<libminisip/conference/ConfMessageRouter.h>
#include<libmutil/itoa.h>
#include<libmutil/Timestamp.h>
#include<libmutil/termmanip.h>
#include<libmutil/dbg.h>
#include<libmsip/SipSMCommand.h>
#include <time.h>
#include<libminisip/gui/LogEntry.h>
#include<libmsip/SipCommandString.h>
#include<libminisip/mediahandler/MediaHandler.h>
#include<libmutil/MemObject.h>
#include <iostream>
#include<time.h>
#include<libmnetutil/IP4Address.h>

#ifdef _WIN32_WCE
#	include"../include/minisip_wce_extra_includes.h"
#endif

using namespace std;


/*
                                                             a13:reject/send40X
                                                                  +------------+
                                                                  |            |
                 +---------------+   INVITE           +--------------+CANCEL   |
                 |               |   a10:transIR; 180 |              |a12:tCncl|
                 |     start     |------------------->|    ringing   |---------+
                 |               |                    |              |         |
a26transport_err +---------------+                    +--------------+         |
gui(failed)              |                                    |                |
  +----------------+     | invite                             |                |
  |                |     V a0: new TransInvite                |                |
  |              +---------------+                            |                |
  |         +----|               |----+                       |                |
  |     1xx |    |Calling_noauth |    | 180                   |                |
  |a2:(null)+--->|               |<---+ a1: gui(ringing)      |                |
  +--------------+---------------+                            |                |
  |                X   ^ |     |       2xx/a3: send ACK       |                |
  |    [XACK/a15X] +---+ |     +------------------------------+                |
  |                      |                                    |                |
  +----------------+     |  40X                               | accept_invite  |
  |                |     V  a20: send_auth                    | a11: send 200  |
  |              +---------------+                            |                |
  |         +----|               |----+                       |                |
  |     1xx |    |Calling_stored |    | 180                   |                |
  |a22:(nul)+--->|               |<---+ a21: gui(ringing)     |                |
  |              +---------------+                            |                |
  |                      |                                    |                |
  |                      | 2xx                                |                |
  |                      v a23: [Xsend ACKX]                  |                |
  |              +---------------+                            |                |
  |          +---|               |<---------------------------+                |
  |  ACK            |   in call     |-------+                                     |
  |  a27     +---|               |       |                                     |
  |              +---------------+       |                                     |
  |                      |               | bye                                 |
  |cancel                | BYE           | a6:TransByeInit                     |
  |a8/a: new TrnsCncl    V a5:new ByeResp|                                     |
  +------------->+---------------+       |                                     |
  |3-699/a9:gui()|               |       |                                     |
  +------------->|   termwait    |<------+-------------------------------------+
  |              |               |
  +------------->+---------------+
                         |
			 | a25: no_transactions
			 V phone(call_terminated)
                 +---------------+
		 |               |
		 |  terminated   |
		 |               |
                 +---------------+
		 
   CANCEL
   a7:TrnsCnclRsp
   
*/
 
 
bool SipDialogConfVoip::a0_start_callingnoauth_invite( const SipSMCommand &command)
{//cerr<<"************************************invite receivedqwe"<<endl;
	if (transitionMatch(command, 
				SipCommandString::invite,
				SipSMCommand::dialog_layer,
				SipSMCommand::dialog_layer)){
#ifdef ENABLE_TS
		ts.save("a0_start_callingnoauth_invite");
#endif
		//cerr<<"sending invite1*************"<<endl;
		//int seqNo = requestSeqNo();
		//cerr<<"dialogState.seqNo*************"+itoa(dialogState.seqNo)<<endl;
		++dialogState.seqNo;
//		setLocalCalled(false);
		localCalled=false;
		dialogState.remoteUri= command.getCommandString().getParam();
		//cerr<<"dialogState.callId*************"+dialogState.callId<<endl;
		//cerr<<"dialogState.seqNo*************"+itoa(dialogState.seqNo)<<endl;

/*		MRef<SipTransaction*> invtrans = new SipTransactionInviteClient(sipStack, 
				//MRef<SipDialog *>(this), 
				dialogState.seqNo, 
				"INVITE", 
				dialogState.callId);
		
		invtrans->setSocket( phoneconf->proxyConnection );
		//cerr<<"sending invite3*************"<<endl;
		dispatcher->getLayerTransaction()->addTransaction(invtrans);
		//registerTransactionToDialog(invtrans);
*/
		//cerr<<"sending invite*************"<<endl;
		/*CommandString cmdstr("", "myuri", getDialogConfig()->inherited.sipIdentity->getSipUri());
		
		cmdstr.setParam3(confId);
		
		//getDialogContainer()->getCallback()->sipcb_handleCommand(cmdstr);
		getDialogContainer()->getCallback()->sipcb_handleConfCommand( cmdstr );*/
		sendInvite(""/*invtrans->getBranch()*/);
		//MessageRouter * ptr=(MessageRouter *)(getDialogContainer()->getCallback());
		//adviceList=(ptr->getConferenceController()->getConnectedList());
		return true;
	}else{
		return false;
	}

}

bool SipDialogConfVoip::a1_callingnoauth_callingnoauth_18X( const SipSMCommand &command)
{	
	if (transitionMatch(SipResponse::type, command, IGN, SipSMCommand::dialog_layer, "18*")){

	    MRef<SipResponse*> resp= (SipResponse*) *command.getCommandPacket();

#ifdef ENABLE_TS
	    ts.save( RINGING );
#endif
	    CommandString cmdstr(dialogState.callId, SipCommandString::remote_ringing);
	    cmdstr.setParam3(confId);
	    //getDialogContainer()->getCallback()->sipcb_handleCommand(cmdstr);
	    sipStack->getCallback()->handleCommand("sip_conf", cmdstr );

	    //We must maintain the dialog state. We copy the remote tag and
	    //the remote sequence number
	    dialogState.remoteTag = command.getCommandPacket()->getHeaderValueTo()->getParameter("tag");
	    dialogState.remoteSeqNo = command.getCommandPacket()->getCSeq();
	    
	    string peerUri = command.getCommandPacket()->getFrom().getString();

	    MRef<SdpPacket*> sdp((SdpPacket*)*resp->getContent());
	    if ( !sdp.isNull() ){
		    //Early media
		    getMediaSession()->setSdpAnswer( sdp, peerUri );
	    }

	    return true;
	}
	else{
	    return false;
	}
}


bool SipDialogConfVoip::a2_callingnoauth_callingnoauth_1xx( const SipSMCommand &command)
{

	if (transitionMatch(SipResponse::type, command, IGN, SipSMCommand::dialog_layer, "1**")){
		dialogState.remoteTag = command.getCommandPacket()->getHeaderValueTo()->getParameter("tag");
		return true;
	}else{
		return false;
	}
    
}

bool SipDialogConfVoip::a3_callingnoauth_incall_2xx( const SipSMCommand &command)
{
	if (transitionMatch(SipResponse::type, command, IGN, SipSMCommand::dialog_layer, "2**")){
#ifdef ENABLE_TS
		ts.save("a3_callingnoauth_incall_2xx");
#endif

		MRef<SipResponse*> resp(  (SipResponse*)*command.getCommandPacket() );
		lastResponse=resp;
		string peerUri = resp->getFrom().getString();
		setLogEntry( new LogEntryOutgoingCompletedCall() );
		getLogEntry()->start = time( NULL );
		getLogEntry()->peerSipUri = peerUri;

		dialogState.remoteTag = command.getCommandPacket()->getHeaderValueTo()->getParameter("tag");
		//parsing sdp part for conf:
		massert(dynamic_cast<SdpPacket*>(*resp->getContent())!=NULL);
		MRef<SdpPacket*> sdp = (SdpPacket*)*resp->getContent();
		string numToConnect = sdp->getSessionLevelAttribute("conf_#participants");
		int num = 0;

   //--- Convert each digit char and add into result.
   			int t=0;
			while (numToConnect[t] >= '0' && numToConnect[t] <='9') {
      				num = (num * 10) + (numToConnect[t] - '0');
      				t++;
   			}
			string users="";
			for(t=0;t<num;t++)
				//connectList[t]=  sdp->getSessionLevelAttribute("participant_"+itoa(t+1));
				users=users+sdp->getSessionLevelAttribute("participant_"+itoa(t+1))+";";
			//cerr<<"==============users: "+users<<endl;	
                
		
		CommandString cmdstr(dialogState.callId, SipCommandString::invite_ok, users,(getMediaSession()->isSecure()?"secure":"unprotected"),confId);
		
		
		
		//getDialogContainer()->getCallback()->sipcb_handleCommand(cmdstr);
		sipStack->getCallback()->handleCommand("sip_conf", cmdstr );
		//cerr<<"****************************sendack is called**********************"<<endl;
		//sendAck(getLastInvite()->getDestinationBranch() );
		//BM
		MRef<ConfMessageRouter*> ptr= confCallback;//(ConfMessageRouter*) *(getDialogContainer()->getConfCallback());
		adviceList=(ptr->getConferenceController(confId)->getConnectedList());
		sendAck(getLastInvite()->getDestinationBranch());
		
		if(!sortMIME(*resp->getContent(), peerUri, 3))
			return false;
			
			
#ifdef IPSEC_SUPPORT
		// Check if IPSEC was required
		if (ipsecSession->required() && !ipsecSession->offered)
			return false;
#endif
		return true;
	}else{
		return false;
	}
}

bool SipDialogConfVoip::a5_incall_termwait_BYE( const SipSMCommand &command)
{
	
	if (transitionMatch("BYE", command, SipSMCommand::transaction_layer, SipSMCommand::dialog_layer)){
		MRef<SipRequest*> bye = (SipRequest*) *command.getCommandPacket();

		//mdbg << "log stuff"<< end;
		if( getLogEntry() ){
			((LogEntrySuccess *)(*( getLogEntry() )))->duration = 
			time( NULL ) - getLogEntry()->start; 

			getLogEntry()->handle();
		}

/*
		MRef<SipTransaction*> byeresp = new SipTransactionNonInviteServer(sipStack, 
				//MRef<SipDialog*>(this), 
				bye->getCSeq(),
				bye->getLastViaBranch(), 
				bye->getCSeqMethod(), 
				dialogState.callId);

		dispatcher->getLayerTransaction()->addTransaction(byeresp);
		//registerTransactionToDialog(byeresp);

		SipSMCommand cmd(command);
		cmd.setDestination(SipSMCommand::transaction_layer);
		cmd.setSource(command.getSource());
		dispatcher->enqueueCommand(cmd, HIGH_PRIO_QUEUE);
*/		
		sendByeOk(bye, ""/*byeresp->getBranch()*/ );

		CommandString cmdstr(dialogState.callId, SipCommandString::remote_hang_up);
		cmdstr.setParam3(confId);
		//getDialogContainer()->getCallback()->sipcb_handleCommand(cmdstr);
		sipStack->getCallback()->handleCommand("sip_conf", cmdstr );

		getMediaSession()->stop();

		
		signalIfNoTransactions();
		return true;
	}else{
		return false;
	}
}


bool SipDialogConfVoip::a6_incall_termwait_hangup( const SipSMCommand &command)
{
//	merr << "EEEEEEEE: a6: got command."<< end;
//	merr <<"EEEEE: type is "<< command.getType()<< end;
	if (transitionMatch(command, 
				SipCommandString::hang_up,
				SipSMCommand::dialog_layer,
				SipSMCommand::dialog_layer)){
//		merr << "EEEEEE match"<< end;
//		setCurrentState(toState);
		//int bye_seq_no= requestSeqNo();
		++dialogState.seqNo;
/*		MRef<SipTransaction*> byetrans( new SipTransactionNonInviteClient(sipStack, 
					//MRef<SipDialog*>(this), 
					dialogState.seqNo, 
					"BYE", 
					dialogState.callId)); 

		dispatcher->getLayerTransaction()->addTransaction(byetrans);
		//registerTransactionToDialog(byetrans);
*/
		sendBye(""/*byetrans->getBranch()*/, dialogState.seqNo);
		
		if (getLogEntry()){
			(dynamic_cast< LogEntrySuccess * >(*( getLogEntry() )))->duration = time( NULL ) - getLogEntry()->start; 
			getLogEntry()->handle();
		}
		
		getMediaSession()->stop();

		signalIfNoTransactions();
		return true;
	}else{
//		merr << "EEEEEE no match"<< end;
		return false;
	}
}

bool SipDialogConfVoip::a7_callingnoauth_termwait_CANCEL( const SipSMCommand &command)
{
	if (transitionMatch("CANCEL", command, SipSMCommand::transaction_layer, IGN)){
//		setCurrentState(toState);

/*		MRef<SipTransaction*> cancelresp( 
				new SipTransactionNonInviteServer(
					sipStack,