sipdialogvoip.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
 * 	SipDialogVoip.cxx
 * Author
 * 	Erik Eliasson, [e-mail]
 * Purpose
 * 
*/
#include<config.h>

#include<libminisip/sip/SipDialogVoip.h>

#include<libmutil/massert.h>

//#include<libmsip/SipDialogContainer.h>
#include<libmsip/SipTransactionInviteClientUA.h>
#include<libmsip/SipTransactionInviteServerUA.h>
#include<libmsip/SipTransactionNonInviteClient.h>
#include<libmsip/SipTransactionNonInviteServer.h>
#include<libmsip/SipTransactionUtils.h>
#include<libmsip/SipCommandString.h>
#include<libmsip/SipHeaderWarning.h>
#include<libmsip/SipHeaderContact.h>
#include<libmsip/SipHeaderFrom.h>
#include<libmsip/SipHeaderRoute.h>
#include<libmsip/SipHeaderRequire.h>
#include<libmsip/SipHeaderTo.h>
#include<libmsip/SipMIMEContent.h>
#include<libmsip/SipMessageContent.h>
#include<libmutil/itoa.h>
#include<libmcrypto/base64.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<libmutil/print_hex.h>
#include <iostream>
#include<time.h>

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

using namespace std;


/*
 
 
  
States are named as follows where X and NN is digit numbers

  a 1 X NN _ "fromstate" _ "tostate" _ "trigger"
    ^ ^ ^^
    | | ++-- Number starting from 1
    | +----- Type of functionality 0: in-call commands such as BYE, bye and re-invite
    |                              1: dialog management 
    |                              2: locally initiated call transfer 
    |                              3: remotely initiated call transfer
    +------- Actions in this file. SipDialogVoipClient=2, SipDialogVoipServer=3
    
Example: a1301_incall_transfaskuser_REFER means that it is the transition
         from the incall state to the transfaskuser state that will happen
	 if it receives a REFER. a1301 means "action 1 in SipDialogVoip that
	 handles remotely initiated call transfer".

	 
State Machine:
 The state machine in this file only deals with a call after it has been set up and
 is in the "in_call" state. Sub-classes implement call setup so that we get into
 this state.
 



 

  These states are              These states are              These states
  when the local user           for a "normal" call.          are used when a
  requests a call               in_call is during             remote user request
  transfer.                     the call.                     call transfer
 <---------------->            <--------------------->        <---------------->
                   transfer
                   a1201:new TReferInit           REFER
 +---------------+              +---------------+ a1301:      +---------------+
 |               |<-------------|               |---------->|               |
 | transf_request|              |   in call     |refus/a1303|transf_ask_user|
 |               |------------->|               |<----------|               |
 +---------------+ 300-699      +---------------+-------+   +---------------+
         |         a1203: -             |               | hangup          | accept_transfer
         | 202                          | BYE           | a1002:TByeCli   | a1302: 202;start new call
         | a1202: new subscription      V a1001:TByeServ|                 V
 +---------------+              +---------------+       |   +---------------+
 |               |------------->|               |<------+   |               |
 | transf_pending|  bye         |   bye_request |           |transf_started |
 |               |  a1001   +---|               |<----------|               |
 +---------------+          |   +---------------+  bye      +---------------+
   |          ^ |           |      ^    |          a1304             |     
   |          | | hang_up   +------+    |  a1003: 2XX-6XX            |        
   +----------+ | a1102: call_term_earlyV  gui(call_terminated)      |    
     NOTIFY     |               +---------------+                    |
     a1204: -   |               |               |                    |
                +-------------->|   termwait    |<-------------------+
                BYE             |               |         BYE
                a1002 gui(cterm)+---------------+         a1305 gui(call_terminated)
                                 v      |
				a1102 	| no_transactions
					V a1304
                                +---------------+
                                |               |
                                |  terminated   |
                                |               |
                                +---------------+
 
					
  
 Why do we inform the GUI that the dialog has ben terminated (in a1101)? --Erik
 
*/


/**
 * Helper function that returns the uri in the referto header.
 */
static string getReferredUri(MRef<SipRequest*> req){
	string referredUri;
	MRef<SipHeaderValue*> hval;
	if (req)
		hval = req->getHeaderValueNo(SIP_HEADER_TYPE_REFERTO, 0);

	if (hval)
		referredUri = hval->getString();
	else{
		cerr << "WARNING: Referred to uri not found!"<<endl;
	}
	return referredUri;
}

/**
 *
 * This transaction is also used a a1001_transfpending_termwait_BYE
 */
bool SipDialogVoip::a1001_incall_termwait_BYE( const SipSMCommand &command)
{
	
	if (transitionMatch("BYE", 
				command, 
				SipSMCommand::transaction_layer, 
				SipSMCommand::dialog_layer) &&
	    dialogState.remoteTag != ""){
		MRef<SipRequest*> bye = (SipRequest*) *command.getCommandPacket();

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

			getLogEntry()->handle();
		}

#if 0
		MRef<SipTransaction*> byeresp = new SipTransactionNonInviteServer(sipStack, 
				//MRef<SipDialog*>(this), 
				bye->getCSeq(),
				bye->getCSeqMethod(),
				bye->getLastViaBranch(), 
				dialogState.callId); //TODO: remove second argument

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

		SipSMCommand cmd(command);
		cmd.setDestination(SipSMCommand::transaction_layer);
		cmd.setSource(command.getSource());
		dispatcher->enqueueCommand(cmd, HIGH_PRIO_QUEUE/*, PRIO_LAST_IN_QUEUE*/);
#endif

		sendByeOk(bye, /*byeresp->getBranch()*/"" );

		CommandString cmdstr(dialogState.callId, SipCommandString::remote_hang_up);
		sipStack->getCallback()->handleCommand("gui", cmdstr);

		getMediaSession()->stop();

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


#if 0
bool SipDialogVoip::a1001_incall_termwait_BYE( const SipSMCommand &command)
{
	
	if (transitionMatch("BYE", 
				command, 
				SipSMCommand::transaction_layer, 
				SipSMCommand::dialog_layer) &&
	    dialogState.remoteTag != ""){
		MRef<SipRequest*> bye = (SipRequest*) *command.getCommandPacket();

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

			getLogEntry()->handle();
		}

#if 0
		MRef<SipTransaction*> byeresp = new SipTransactionNonInviteServer(sipStack, 
				//MRef<SipDialog*>(this), 
				bye->getCSeq(),
				bye->getCSeqMethod(),
				bye->getLastViaBranch(), 
				dialogState.callId); //TODO: remove second argument

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

		SipSMCommand cmd(command);
		cmd.setDestination(SipSMCommand::transaction_layer);
		cmd.setSource(command.getSource());
		dispatcher->enqueueCommand(cmd, HIGH_PRIO_QUEUE/*, PRIO_LAST_IN_QUEUE*/);
#endif

		sendByeOk(bye, /*byeresp->getBranch()*/"" );

		CommandString cmdstr(dialogState.callId, SipCommandString::remote_hang_up);
		sipStack->getCallback()->handleCommand("gui", cmdstr);

		getMediaSession()->stop();

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


/**
 *
 * This transaction is also used a a1001_transfpending_byerequest_hangup
 */
bool SipDialogVoip::a1002_incall_byerequest_hangup( const SipSMCommand &command)
{
	if (transitionMatch(command, 
				SipCommandString::hang_up,
				SipSMCommand::dialog_layer,
				SipSMCommand::dialog_layer)){
		//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{
		return false;
	}
}

///We consume any final response, and consider the session stopped (even if
///the response is not 2XX)
bool SipDialogVoip::a1003_byerequest_termwait_26( const SipSMCommand &command){
	if (transitionMatch(SipResponse::type, 
				command, 
				SipSMCommand::transaction_layer,
				SipSMCommand::dialog_layer,
				"2**\n3**\n4**\n5**\n6**") ){
/*		SipSMCommand cmd(
				CommandString( dialogState.callId, 
					SipCommandString::call_terminated),
					SipSMCommand::dialog_layer,
					SipSMCommand::dispatcher);

		dispatcher->enqueueCommand( cmd, HIGH_PRIO_QUEUE);
*/
	/* Tell the GUI */
		CommandString cmdstr( dialogState.callId, SipCommandString::call_terminated);
		sipStack->getCallback()->handleCommand("gui", cmdstr );

#ifdef IPSEC_SUPPORT
		if(ipsecSession){
			cerr << "Clearing" << endl;
			if(ipsecSession->stop() != 0)
				cerr << "Not all IPSEC parameters were confired cleared. Check and remove manually." << endl;