opswapcalls.cxx

Vovida 社区开源的 SIP 协议源码

/* ====================================================================
 * The Vovida Software License, Version 1.0 
 * 
 * Copyright (c) 2000 Vovida Networks, Inc.  All rights reserved.
 * 
 * Redistribution and use in source and binary forms, with or without
 * modification, are permitted provided that the following conditions
 * are met:
 * 
 * 1. Redistributions of source code must retain the above copyright
 *    notice, this list of conditions and the following disclaimer.
 * 
 * 2. Redistributions in binary form must reproduce the above copyright
 *    notice, this list of conditions and the following disclaimer in
 *    the documentation and/or other materials provided with the
 *    distribution.
 * 
 * 3. The names "VOCAL", "Vovida Open Communication Application Library",
 *    and "Vovida Open Communication Application Library (VOCAL)" must
 *    not be used to endorse or promote products derived from this
 *    software without prior written permission. For written
 *    permission, please contact [e-mail].
 *
 * 4. Products derived from this software may not be called "VOCAL", nor
 *    may "VOCAL" appear in their name, without prior written
 *    permission of Vovida Networks, Inc.
 * 
 * THIS SOFTWARE IS PROVIDED "AS IS" AND ANY EXPRESSED OR IMPLIED
 * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES
 * OF MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE, TITLE AND
 * NON-INFRINGEMENT ARE DISCLAIMED.  IN NO EVENT SHALL VOVIDA
 * NETWORKS, INC. OR ITS CONTRIBUTORS BE LIABLE FOR ANY DIRECT DAMAGES
 * IN EXCESS OF $1,000, NOR FOR ANY INDIRECT, INCIDENTAL, SPECIAL,
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 * PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY
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 * USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH
 * DAMAGE.
 * 
 * ====================================================================
 * 
 * This software consists of voluntary contributions made by Vovida
 * Networks, Inc. and many individuals on behalf of Vovida Networks,
 * Inc.  For more information on Vovida Networks, Inc., please see
 * <http://www.vovida.org/>.
 *
 */

static const char* const OpSwapCalls_cxx_Version =
    "$Id: OpSwapCalls.cxx,v 1.30 2002/11/12 20:25:13 veer Exp $";

#include "global.h"
#include "OpSwapCalls.hxx"
#include "UaDeviceEvent.hxx"
#include "UaCallInfo.hxx"
#include "UaStateMachine.hxx"
#include "UaHardwareEvent.hxx"
#include "UaDevice.hxx"
#include "UaConfiguration.hxx"
#include "Sdp2Session.hxx"
#include "Sdp2Media.hxx"
#include "SipRecordRoute.hxx"
#include "SipVia.hxx"

using namespace Vocal;


using Vocal::SDP::SdpSession;
using Vocal::SDP::SdpMedia;
using Vocal::SDP::MediaAttributes;
using Vocal::SDP::ValueAttribute;
using Vocal::SDP::SdpRtpMapAttribute;


OpSwapCalls::OpSwapCalls()
{
}


OpSwapCalls::~OpSwapCalls()
{
}


const char* const
OpSwapCalls::name() const
{
    return "OpSwapCalls";
}


const Sptr < State >
OpSwapCalls::process( const Sptr < SipProxyEvent > event )
{
    cpLog( LOG_DEBUG, "OpSwapCalls operation" );

    Sptr < UaDeviceEvent > deviceEvent;
    deviceEvent.dynamicCast( event );
    if ( deviceEvent == 0 )
    {
        return 0;
    }

    if ( deviceEvent->type != DeviceEventFlash )
    {
        return 0;
    }

    Sptr < SipCallId > call2Id = UaDevice::instance()->getCallWaitingId();
    if ( call2Id != 0 )
    {
        // don't initiate second call if there is a cal on call waiting
        return 0;
    }

    cpLog( LOG_DEBUG, "Swap the call" );

    if ( !putContactOnhold( deviceEvent ) )
    {
        cpLog( LOG_WARNING, "Cannot put contact on hold" );
        return 0;
    }

    restoreHeldCall( deviceEvent );

    return 0;
}


bool
OpSwapCalls::putContactOnhold( const Sptr < UaDeviceEvent > deviceEvent )
{
    Sptr < UaCallInfo > call;
    call.dynamicCast( deviceEvent->getCallInfo() );
    assert( call != 0 );

    //put the current contact on hold
    Sptr < Contact > contact = call->getContact();
    if ( contact == 0 )
    {
        cpLog( LOG_ERR, "No current contact");
        return false;
    }
    Sptr < InviteMsg > reInvite;

    //get the current invite msg
    Sptr < InviteMsg > inviteMsg = call->getRingInvite();
    // the following comparison statement is equavalent to 
    // if ( *inviteMsg != contact->getInviteMsg() )
    // != operator is not used because it is not define.
    if ( !(*inviteMsg == contact->getInviteMsg()) )
    {
        cpLog( LOG_DEBUG, "Putting the 2nd call on hold" );
        firstCall = true;
    }
    else
    {
        cpLog( LOG_DEBUG, "Putting the 1st call on hold" );
        firstCall = false;
    }

    int status = contact->getStatus();
    if ( status == 200 )
    {
        cpLog( LOG_DEBUG, "Put callee on hold" );

        const StatusMsg& msg = contact->getStatusMsg();
        if ( &msg != 0 )
        {
            reInvite = new InviteMsg( msg );

            //add SDP
            Sptr < SipSdp > localSdp = call->getLocalSdp();
            assert( localSdp != 0 );
            SipSdp sipSdp = *localSdp;
            reInvite->setContentData( &sipSdp );
        }
        else
        {
            cpLog( LOG_ERR , "No status message for re-INVITE" );
        }
    }
    else
    {
        cpLog( LOG_DEBUG, "Put caller on hold" );

        const InviteMsg& msg = contact->getInviteMsg();
        if ( &msg != 0 )
        {
            string sipPort = UaConfiguration::instance()->getLocalSipPort();
            reInvite = new InviteMsg( msg.getFrom().getUrl(),
                                      atoi( sipPort.c_str() ) );
            SipFrom from( msg.getTo().getUrl() );
            reInvite->setFrom( from );
            //TODO Check if it is necessary to set To:
            reInvite->setCallId( msg.getCallId() );

            // Convert RecordRoute to reverse Route
            int numRecordRoute = msg.getNumRecordRoute();
            SipRecordRoute recordroute;
            SipRoute route;

            for ( int i = 0; i < numRecordRoute; i++ )
            {
                recordroute = msg.getRecordRoute( i );
                route.setUrl( recordroute.getUrl() );
                reInvite->setRoute( route );     // to beginning
            }

            int numContact = msg.getNumContact();
            if ( numContact )
            {
                SipContact contact = msg.getContact( numContact - 1 );
                route.setUrl( contact.getUrl() );
                reInvite->setRoute( route );     // to beginning
            }

        }
        else
        {
            cpLog( LOG_ERR , "No INVITE message for re-INVITE" );
        }
    }
    assert( reInvite != 0 );

    SipVia sipVia;
    sipVia.setprotoVersion( "2.0" );
    sipVia.setHost( Data( theSystem.gethostAddress() ) );
    sipVia.setPort( atoi( UaConfiguration::instance()
			  ->getLocalSipPort().c_str() ) );
    reInvite->flushViaList();
    reInvite->setVia( sipVia, 0 );

    // Set Contact: header
    Sptr< SipUrl > myUrl = new SipUrl;
    myUrl->setUserValue( UaConfiguration::instance()->getUserName(), "phone" );
    myUrl->setHost( Data( theSystem.gethostAddress() ) );
    myUrl->setPort( UaConfiguration::instance()->getLocalSipPort() );
    SipContact me;
    me.setUrl( myUrl );
    reInvite->setNumContact( 0 );    // Clear
    reInvite->setContact( me );

    //TODO Is it going to be a problem if the other side also use the next
    //TODO CSeq at the same time?
    unsigned int cseq = contact->getCSeqNum();
    contact->setCSeqNum( ++cseq );
    SipCSeq sipCSeq = reInvite->getCSeq();
    sipCSeq.setCSeq( cseq );
    reInvite->setCSeq( sipCSeq );

    Sptr<SipSdp> sipSdp;
    sipSdp.dynamicCast ( reInvite->getContentData( 0 ) );

    assert ( sipSdp != 0 );
    SdpSession sdpDesc = sipSdp->getSdpDescriptor();
    sdpDesc.setHold();
    sipSdp->setSdpDescriptor( sdpDesc );