defaultdialoghandler.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]>
*/

#include<config.h>

#include<libminisip/sip/DefaultDialogHandler.h>

#include<libmnetutil/IP4Address.h>
#include<libmnetutil/NetworkException.h>

#include<libmsip/SipDialogRegister.h>
//#include<libmsip/SipDialogContainer.h>

#include<libmsip/SipHeaderFrom.h>
#include<libmsip/SipHeaderTo.h>
#include<libmsip/SipHeaderAcceptContact.h>
#include<libmsip/SipMessage.h>
#include<libmsip/SipMessageContentIM.h>
#include<libmsip/SipCommandString.h>
#include<libmsip/SipTransactionInviteServer.h>
#include<libmsip/SipTransactionInviteServerUA.h>
#include<libmsip/SipTransactionNonInviteServer.h>
#include<libmsip/SipTransactionNonInviteClient.h>
#include<libmutil/massert.h>

#include<libminisip/sip/SipDialogVoipServer.h>
#include<libminisip/sip/SipDialogVoipServer100rel.h>
#include<libminisip/sip/SipDialogConfVoip.h>
#include<libminisip/sip/SipDialogPresenceClient.h>
#include<libminisip/sip/SipDialogPresenceServer.h>
#include<libminisip/conference/ConfMessageRouter.h>

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

#ifdef P2T_SUPPORT
#	include<libminisip/p2t/P2T.h>
#	include<libminisip/p2t/SipDialogP2Tuser.h>
#endif

#include<libminisip/mediahandler/MediaHandler.h>

#ifdef IPSEC_SUPPORT
#	include<../ipsec/MsipIpsecAPI.h>
#endif


#include<libmutil/dbg.h>

using namespace std;

DefaultDialogHandler::DefaultDialogHandler(MRef<SipStack*> stack, 
            //SipDialogConfig &conf,
	    //MRef<SipDialogConfig *> conf,
	    MRef<SipSoftPhoneConfiguration*> pconf,
	    MRef<MediaHandler *>mediaHandler): 
		sipStack(stack),
                //SipDialog(stack, conf),
		phoneconf(pconf),
		mediaHandler(mediaHandler)
{
	outsideDialogSeqNo=1;

//	dialogState.callId = string("DCH_")+itoa(rand())+"@"+getDialogConfig()->inherited->externalContactIP;
#ifdef P2T_SUPPORT
	//Initialize GroupListServer
	grpListServer=NULL;
#endif
}

DefaultDialogHandler::~DefaultDialogHandler(){
// 	cerr << "~DefaultDialogHandler" << endl;
}

string DefaultDialogHandler::getName(){
	return "DefaultDialogHandler";
}

bool DefaultDialogHandler::handleCommandPacket( MRef<SipMessage*> pkt){

#if 0
	/* First, check if this is a packet that could not be handled by
	 * any transaction and send 481 response if that is the case */
	if (source==SipSMCommand::transaction_layer && dispatchCount>=2){ // this is the packets second run of handling.
		mdbg << "DefaultCallHandler::handleCommand: Detected dispatched already - sending 481"<< end;

		//FIXME: Check what branch parameter to send.
		MRef<SipResponse*> no_call= new SipResponse("nobranch", 481,"Call Leg/Transaction Does Not Exist", MRef<SipMessage*>(*pkt));
		MRef<SipMessage*> pref(*no_call);

		sipStack->getDispatcher()->getLayerTransport()->sendMessage(pref,
				string(""), //branch
				false
				);

		return true;
	}

#ifdef DEBUG_OUTPUT
	if (source==SipSMCommand::transaction_layer&& dispatchCount>=2){ // this is the packets second run of handling.
		cerr << "WARNING: INTERNAL ERROR: command was not handled (dispatched flag indication)"<<endl;
		return true;
	}
#endif
	
#endif

	if (pkt->getType()=="INVITE"){
		//type casting
		//MRef<SipRequest*> inv = MRef<SipRequest*>((SipRequest*)*pkt);
		MRef<SipRequest*> inv = dynamic_cast<SipRequest*>(*pkt);
		//inv->checkAcceptContact();
		//check if it's a regular INVITE or a P2T INVITE
#ifdef P2T_SUPPORT
		if(inv->is_P2T()) {
			inviteP2Treceived(SipSMCommand(pkt,source,destination));	
		}
#endif


		bool isConfJoin=false;
		bool isP2T=false;
		bool isConfConnect=false;
		MRef<SipHeaderValueAcceptContact*> acp;
		int i=0;
		MRef<SipHeaderValue*> hdr=inv->getHeaderValueNo(SIP_HEADER_TYPE_ACCEPTCONTACT, i);
		do{
			if (hdr){
				MRef<SipHeaderValueAcceptContact*> acp = (SipHeaderValueAcceptContact*)*hdr;
				if(acp && acp->getFeaturetag()=="+sip.p2t=\"TRUE\"")
					isP2T=true;
				else if(acp && acp->getFeaturetag()=="+sip.confjoin=\"TRUE\"") {
					//cout << "SIPINVITE: Setting conjoin to true" << endl;
					isConfJoin=true;
				}
				else if(acp && acp->getFeaturetag()=="+sip.confconnect=\"TRUE\""){
					isConfConnect=true;
				}
			}
			i++;
			hdr = inv->getHeaderValueNo(SIP_HEADER_TYPE_ACCEPTCONTACT, i);
		}while(hdr);

		if(isConfJoin) {
			MRef<SipHeaderValueTo*> to = pkt->getHeaderValueTo();
			string uri;
			MRef<SipIdentity *> id = NULL;

			if( to ){
				id = phoneconf->getIdentity( to->getUri() );
			}
#ifdef DEBUG_OUTPUT			
			mdbg << "DefaultDialogHandler:: creating new SipDialogConfVoip" << end;
#endif			
			//MRef<SipMessage*> pack = command.getCommandPacket();
			//MRef<SipInvite*> inv = MRef<SipInvite*>((SipInvite*)*pack);

			//get the GroupList from the remote GroupListServer
			//MRef<GroupList*>grpList;
			massert(dynamic_cast<SdpPacket*>(*inv->getContent())!=NULL);
			MRef<SdpPacket*> sdp = (SdpPacket*)*inv->getContent();
			string confid = sdp->getSessionLevelAttribute("confId");
			string numToConnect = sdp->getSessionLevelAttribute("conf_#participants");


			//this is a join packet and contains an advice list. The list is created from the
			// packet here in order to send it to the GUI for display.


			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++;
			}
			for(t=0;t<num;t++)
				//connectList[t]=  sdp->getSessionLevelAttribute("participant_"+itoa(t+1));
				connectList.push_back((ConfMember(sdp->getSessionLevelAttribute("participant_"+itoa(t+1)),"")));
			//cerr << "DDH: "+numToConnect[0]<< endl;
			//cerr << "DDH: "+numToConnect[1]<< endl;
			cerr << "DDH: "+itoa(num)<< endl;
			//int num=atoi(numToConnect);
			//string gID = sdp->getSessionLevelAttribute("p2tGroupIdentity");
			//string prot = sdp->getSessionLevelAttribute("p2tGroupListProt");
			// get a session from the mediaHandler
			MRef<Session *> mediaSession = 
				mediaHandler->createSession(phoneconf->securityConfig, *(*phoneconf->inherited), pkt->getCallId() );

			MRef<SipDialogConfig*> callConf = new SipDialogConfig(phoneconf->inherited);
			if( id ){
				cerr << "Got a call from Id " << id->getSipUri() << endl;
				callConf->useIdentity( id, false );
			}

#ifdef IPSEC_SUPPORT
			MRef<MsipIpsecAPI *> ipsecSession = new MsipIpsecAPI(mediaHandler->getExtIP(), phoneconf->securityConfig);

			//MRef<SipDialogVoip*> voipCall = new SipDialogVoip(getDialogContainer(), callConf, 
			//					phoneconf, mediaSession, pkt->getCallId(), ipsecSession ); 
			//BM is it safe to pass the list like this?
			MRef<SipDialog*> voipConfCall( new SipDialogConfVoip(sipStack->getConfCallback(),sipStack, callConf, 
						phoneconf, mediaSession, &connectList,confid, pkt->getCallId(), ipsecSession )); 

#else	

			MRef<SipDialog*> voipConfCall( new SipDialogConfVoip(dynamic_cast<ConfMessageRouter*>(*sipStack->getConfCallback()), sipStack, callConf, 
						phoneconf, mediaSession, &connectList,confid, pkt->getCallId()));
#endif
			sipStack->addDialog(voipConfCall);

			
			SipSMCommand cmd(pkt, SipSMCommand::transaction_layer, SipSMCommand::dialog_layer);
			bool handled = voipConfCall->handleCommand(cmd);

			if (!handled){
				cerr<<"Error: DefaultDialogHandler: VoIP dialog refused to handle the incoming INVITE"<<endl;
			}
			
//			cmd.setDispatchCount(dispatchCount);

//			getDialogContainer()->enqueueCommand(cmd, HIGH_PRIO_QUEUE, PRIO_LAST_IN_QUEUE);
//			mdbg << cmd << end;
		}
		else if(isConfConnect) {
			MRef<SipHeaderValueTo*> to = pkt->getHeaderValueTo();
			string uri;
			MRef<SipIdentity *> id = NULL;

			if( to ){
				id = phoneconf->getIdentity( to->getUri() );
			}
#ifdef DEBUG_OUTPUT			
			mdbg << "DefaultDialogHandler:: creating new SipDialogConfVoip" << end;
#endif			


			massert(dynamic_cast<SdpPacket*>(*inv->getContent())!=NULL);
			MRef<SdpPacket*> sdp = (SdpPacket*)*inv->getContent();
			string confid = sdp->getSessionLevelAttribute("confId");
			MRef<Session *> mediaSession = 
				mediaHandler->createSession(phoneconf->securityConfig, *(*phoneconf->inherited), pkt->getCallId() );

			MRef<SipDialogConfig*> callConf = new SipDialogConfig(phoneconf->inherited);

			if( id ){
				cerr << "Got a call from Id " << id->getSipUri() << endl;
				callConf->useIdentity( id, false );
			}

#ifdef IPSEC_SUPPORT
			MRef<MsipIpsecAPI *> ipsecSession = new MsipIpsecAPI(mediaHandler->getExtIP(), phoneconf->securityConfig);

			//MRef<SipDialogVoip*> voipCall = new SipDialogVoip(getDialogContainer(), callConf, 
			//					phoneconf, mediaSession, pkt->getCallId(), ipsecSession ); 
			//BM is it safe to pass the list like this?
			MRef<SipDialog*> voipConfCall( new SipDialogConfVoip(sipStack, callConf, 
						phoneconf, mediaSession, confid, pkt->getCallId(), ipsecSession )); 

#else	

			MRef<SipDialog*> voipConfCall( new SipDialogConfVoip(dynamic_cast<ConfMessageRouter*>(*sipStack->getConfCallback()),sipStack, callConf, 
						phoneconf, mediaSession, confid, pkt->getCallId()));
#endif
			sipStack->addDialog(voipConfCall);

			SipSMCommand cmd(pkt, SipSMCommand::transaction_layer, SipSMCommand::dialog_layer);
			//cmd.setDispatchCount(dispatchCount);

			sipStack->getDispatcher()->enqueueCommand(cmd, HIGH_PRIO_QUEUE/*, PRIO_LAST_IN_QUEUE*/);
			mdbg << cmd << end;
		}
		//start SipDialogVoIP
		else{
			MRef<SipHeaderValueTo*> to = pkt->getHeaderValueTo();
			string uri;
			MRef<SipIdentity *> id = NULL;

			if( to ){
				id = phoneconf->getIdentity( to->getUri() );
			}

			// get a session from the mediaHandler
			MRef<Session *> mediaSession = 
				mediaHandler->createSession(phoneconf->securityConfig, *(*phoneconf->inherited), pkt->getCallId() );

			MRef<SipDialogConfig*> callConf = new SipDialogConfig(phoneconf->inherited);

			if( id ){
				cerr << "Got a call from Id " << id->getSipUri() << endl;
				callConf->useIdentity( id, false );
			}

#ifdef IPSEC_SUPPORT
			MRef<MsipIpsecAPI *> ipsecSession = new MsipIpsecAPI(mediaHandler->getExtIP(), phoneconf->securityConfig);

			MRef<SipDialog*> voipCall( new SipDialogVoipServer(sipStack, callConf, 
						phoneconf, mediaSession, pkt->getCallId(), ipsecSession )); 

#else	
			MRef<SipDialog*> voipCall;
//			if (pkt->supported("100rel")){
//				voipCall = new SipDialogVoipServer100rel(sipStack,
//						callConf,
//						phoneconf,
//						mediaSession,
//						pkt->getCallId());
//			}else{