sipdialogvoipclient.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) 2004,2005,2006
 *
 * Authors: Erik Eliasson <[e-mail]>
 *          Johan Bilien <[e-mail]>
 *	    Joachim Orrblad <joachim[at]orrblad.com>
*/

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

#include<libminisip/sip/SipDialogVoipClient.h>

#include<libmutil/massert.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;


/*
 This class is responsible for starting up a session and ending up in
 the in_call state.
   
 The "dotted" states are implemented in SipDialogVoip.cxx.
   


                 +---------------+
                 |               |
                 |     start     |
                 |               |
a26transport_err +---------------+      100rel
gui(failed)              |              a2015:PRACK +------------------+
  +----------------+     | invite                   |                  |
  |                |     V a2001: new TransInvite   |    prack_sent    |
  |              +---------------+----------------->|                  |
  |         +----|               |<------------(*)--+------------------+
  |     1xx |    |Calling_noauth |----+ 180
  |a3:(null)+--->|               |    | a2002: gui(ringing)      (*) 200 OK
  +--------------+---------------+<---+                          a2016: -
  |                      |     |       2xx/a2004:
  |   cancel|hangup      |     +------------------------------+
  |   a2014              |                                    |
  +----------------+     |  40X                               | accept_invite
  |                |     V  a2008: send_auth                  | a11: send 200
  |              +---------------+                            |
  |         +----|               |----+                       |
  |         |    |Calling_stored |    | 180                   |
  |    1XX  +--->|               |<---+ a2009: gui(ringing)   |
  |    a2010     +---------------+                            |
  |                      |                                    |
  |                      | 2xx                                |
  |                      v a2011:                             |
  |              +. . . . . . . .+                            |
  |              .               .<---------------------------+
  |              .   in call     .        
  |              .               .        
  |              +. . . . . . . .+        
  |                                           
  |CANCEL/cancel                                                 
  |a2005/a2006: new TrnsCncl                     
  +------------->+. . . . . . . .+        
  |33/a2007:gui()|               .        
  +------------->|   termwait    .        
  |trnsperr/a2013|               .
  +------------->+. . . . . . . .+
                          
                                               
                                                 
                 +. . . . . . . .+
                 .               .
                 .  terminated   .
                 .               .
                 +. . . . . . . .+
   
*/

bool SipDialogVoipClient::a2001_start_callingnoauth_invite( const SipSMCommand &command)
{
	if (transitionMatch(command, 
				SipCommandString::invite,
				SipSMCommand::dialog_layer,
				SipSMCommand::dialog_layer)){
#ifdef ENABLE_TS
		ts.save("a0_start_callingnoauth_invite");
#endif
		++dialogState.seqNo;
		
		//set an "early" remoteUri ... we will update this later
		dialogState.remoteUri= command.getCommandString().getParam();

/*		
		MRef<SipTransaction*> invtrans = new SipTransactionInviteClientUA(sipStack, 
				//MRef<SipDialog *>(this), 
				dialogState.seqNo,
				"INVITE",
				dialogState.callId);
		
		invtrans->setSocket( phoneconf->proxyConnection );
		
 		notifyEarlyTermination = true; // set to true, once sent the first command, set to false
		
		dispatcher->getLayerTransaction()->addTransaction(invtrans);
		//registerTransactionToDialog(invtrans);
*/
		sendInvite(""/*invtrans->getBranch()*/);

		return true;
	}else{
		return false;
	}
}

bool SipDialogVoipClient::a2002_callingnoauth_callingnoauth_18X( const SipSMCommand &command)
{	
	if (transitionMatch(SipResponse::type, 
				command, 
				SipSMCommand::transaction_layer, 
				SipSMCommand::dialog_layer, 
				"18*")){

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

#ifdef ENABLE_TS
		ts.save( RINGING );
#endif
		CommandString cmdstr(dialogState.callId, SipCommandString::remote_ringing);
		sipStack->getCallback()->handleCommand("gui", cmdstr);
	
		//We must maintain the dialog state. 
		dialogState.updateState( resp );
		
		//string peerUri = command.getCommandPacket()->getTo().getString();
		string peerUri = dialogState.remoteUri; //use the dialog state ...

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

		return true;
	}else{
		return false;
	}
}


bool SipDialogVoipClient::a2003_callingnoauth_callingnoauth_1xx( const SipSMCommand &command)
{

	if (transitionMatch(SipResponse::type, command, SipSMCommand::transaction_layer, SipSMCommand::dialog_layer, "1**")){
		dialogState.updateState( MRef<SipResponse*>((SipResponse *)*command.getCommandPacket()) );
		return true;
	}else{
		return false;
	}
    
}

bool SipDialogVoipClient::a2004_callingnoauth_incall_2xx( const SipSMCommand &command)
{
	if (transitionMatch(SipResponse::type, 
				command, 
				SipSMCommand::transaction_layer, 
				SipSMCommand::dialog_layer, 
				"2**")){
#ifdef ENABLE_TS
		ts.save("a3_callingnoauth_incall_2xx");
#endif
		MRef<SipResponse*> resp(  (SipResponse*)*command.getCommandPacket() );
		
		string peerUri = dialogState.remoteUri;
		if(!sortMIME(*resp->getContent(), peerUri, 3))
			return false;

		dialogState.updateState( resp );
		
		//string peerUri = resp->getFrom().getString();
		
		setLogEntry( new LogEntryOutgoingCompletedCall() );
		getLogEntry()->start = time( NULL );
		getLogEntry()->peerSipUri = peerUri;


//FIXME: CESC: for now, route set is updated at the transaction layer		
		
		CommandString cmdstr(dialogState.callId, SipCommandString::invite_ok, "",
							(getMediaSession()->isSecure()?"secure":"unprotected"));
		
		sipStack->getCallback()->handleCommand("gui", cmdstr);
		
#ifdef IPSEC_SUPPORT
		// Check if IPSEC was required
		if (ipsecSession->required() && !ipsecSession->offered)
			return false;
#endif
		return true;
	}else{
		return false;
	}
}

bool SipDialogVoipClient::a2005_callingnoauth_termwait_CANCEL( const SipSMCommand &command)
{
	if (transitionMatch("CANCEL", command, SipSMCommand::transaction_layer, SipSMCommand::dialog_layer)){
//		setCurrentState(toState);

/*		MRef<SipTransaction*> cancelresp( 
				new SipTransactionNonInviteServer(
					sipStack,
					//MRef<SipDialog*>(this), 
					command.getCommandPacket()->getCSeq(), 
					command.getCommandPacket()->getCSeqMethod(), 
					command.getCommandPacket()->getLastViaBranch(), 
					dialogState.callId ));

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


		SipSMCommand cmd(command);
		cmd.setSource(SipSMCommand::dialog_layer);
		cmd.setDestination(SipSMCommand::transaction_layer);
		dispatcher->enqueueCommand(cmd, HIGH_PRIO_QUEUE);
*/
		getMediaSession()->stop();
		signalIfNoTransactions();
		return true;
	}else{
		return false;
	}
}

bool SipDialogVoipClient::a2006_callingnoauth_termwait_cancel( const SipSMCommand &command)
{
	if (		transitionMatch(command, 
				SipCommandString::cancel,
				SipSMCommand::dialog_layer,
				SipSMCommand::dialog_layer) 
			|| transitionMatch(command, 
				SipCommandString::hang_up,
				SipSMCommand::dialog_layer,
				SipSMCommand::dialog_layer)){
//		setCurrentState(toState);

/*
		string inv_branch = getLastInvite()->getFirstViaBranch();
		MRef<SipTransaction*> canceltrans( 
				new SipTransactionNonInviteClient(sipStack, 
					//MRef<SipDialog*>( this ), 
					dialogState.seqNo, 
					"CANCEL", 
					dialogState.callId)); 

		canceltrans->setBranch(inv_branch);
		dispatcher->getLayerTransaction()->addTransaction(canceltrans);
		//registerTransactionToDialog(canceltrans);
*/
 		sendCancel(""/*inv_branch*/);

		getMediaSession()->stop();
		signalIfNoTransactions();
		return true;
	}else{
		return false;
	}
}

//Note: This is also used as: callingauth_terminated_36
bool SipDialogVoipClient::a2007_callingnoauth_termwait_36( const SipSMCommand &command)
{
	if (transitionMatch(SipResponse::type, command, SipSMCommand::transaction_layer, SipSMCommand::dialog_layer, "3**\n4**\n5**\n6**")){
		
		MRef<LogEntry *> rejectedLog( new LogEntryCallRejected() );
		rejectedLog->start = time( NULL );
		rejectedLog->peerSipUri = dialogState.remoteTag;
		
		if (sipResponseFilterMatch(MRef<SipResponse*>((SipResponse*)*command.getCommandPacket()),"404")){