opbeingxferred.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,
 * EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
 * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
 * PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY
 * OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
 * (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE
 * 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 OpBeingXferred_cxx_Version =
    "$Id: OpBeingXferred.cxx,v 1.30 2002/11/12 20:25:13 veer Exp $";

#include "global.h"
#include "SipEvent.hxx"
#include "TransferMsg.hxx"

#include "OpBeingXferred.hxx"
#include "UaCallInfo.hxx"
#include "UaConfiguration.hxx"
#include "UaDevice.hxx"
#include "Sdp2Session.hxx"
#include "Sdp2Connection.hxx"
#include "Sdp2Media.hxx"
#include "SipTransferTo.hxx"
#include "SipVia.hxx"

using namespace Vocal;


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


OpBeingXferred::OpBeingXferred()
{
}


OpBeingXferred::~OpBeingXferred()
{
}


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


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

    Sptr < SipEvent > sipEvent;
    sipEvent.dynamicCast( event );
    assert( event != 0 );

    Sptr < SipMsg > sipMsg = sipEvent->getSipMsg();
    assert( sipMsg != 0 );

    Sptr < TransferMsg > msg;
    msg.dynamicCast( sipMsg );
    assert( msg != 0 );

    Sptr < UaCallInfo > call;
    call.dynamicCast( event->getCallInfo() );
    assert( call != 0 );

    //save the transfer msg;
    call->setXferMsg(msg);

    //get to person to be tranferred to
    SipTransferTo xferTo = msg->getTransferTo();

    //form an invite to that person
    string sipPort = UaConfiguration::instance()->getLocalSipPort();

    Sptr< BaseUrl > baseUrl = xferTo.getUrl();
    assert( baseUrl != 0 );
    if( baseUrl->getType() == TEL_URL )
    {
	cpLog( LOG_ERR, "TEL_URL currently not supported\n" );
	assert( 0 );
    }
    // Assume we have a SIP_URL
    Sptr< SipUrl > xferUrl;
    xferUrl.dynamicCast( baseUrl );
    assert( xferUrl != 0 );

    InviteMsg inviteMsg( xferUrl, atoi( sipPort.c_str() ) );

    SipCallId callId;
    callId = call->getContact()->getInviteMsg().getCallId();
    inviteMsg.setCallId( callId );

    SipTo to = inviteMsg.getTo();

    baseUrl = to.getUrl();
    assert( baseUrl != 0 );
    if( baseUrl->getType() == TEL_URL )
    {
	cpLog( LOG_ERR, "TEL_URL currently not supported\n" );
	assert( 0 );
    }
    // Assume we have a SIP_URL
    Sptr< SipUrl > toUrl;
    toUrl.dynamicCast( baseUrl );
    assert( toUrl != 0 );

    to.setDisplayName( xferTo.getDisplayName() );
    inviteMsg.setTo( to );


    //force invite to go through proxy server if a proxy exists
    string proxyUrlStr = UaConfiguration::instance()->getProxyServer();

    if ( proxyUrlStr.length() > 0 )
    {
        Data urlStr = Data( string("sip:") + proxyUrlStr );
        Sptr< SipUrl > proxyUrl = new SipUrl( urlStr );
        SipRequestLine msRequestLine = inviteMsg.getRequestLine();

	baseUrl = msRequestLine.getUrl();
	assert( baseUrl != 0 );
	// Assume we have a SIP_URL
	Sptr< SipUrl > msUrl;
	msUrl.dynamicCast( baseUrl );
	assert( msUrl != 0 );

        msUrl->setHost( proxyUrl->getHost() );
        msUrl->setPort( proxyUrl->getPort() );

        msRequestLine.setUrl( msUrl );
        inviteMsg.setRequestLine( msRequestLine );

        toUrl->setHost( proxyUrl->getHost() );
        toUrl->setPort( proxyUrl->getPort() );
    }

    SipFrom from = inviteMsg.getFrom();
    from.setUser( Data( UaConfiguration::instance()->getUserName() ) );
    from.setDisplayName( Data( UaConfiguration::instance()->getDisplayName() ) );
    inviteMsg.setFrom( from );

    // Set transport in Via:
    SipVia via = inviteMsg.getVia();
    inviteMsg.removeVia();
    via.setTransport( UaConfiguration::instance()->getSipTransport() );
    inviteMsg.setVia( via );

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

    Sptr<SipSdp> sipSdp;
    sipSdp.dynamicCast ( inviteMsg.getContentData( 0 ) );
    if ( sipSdp != 0 )
    {
        SdpSession sdpDesc = sipSdp->getSdpDescriptor();
        list < SdpMedia* > mediaList;
        mediaList = sdpDesc.getMediaList();
        list < SdpMedia* > ::iterator mediaIterator = mediaList.begin();
        MediaAttributes* mediaAttrib;
        mediaAttrib = (*mediaIterator)->getMediaAttributes();
        if ( mediaAttrib == 0 )
        {
            mediaAttrib = new MediaAttributes();
            assert( mediaAttrib );
            (*mediaIterator)->setMediaAttributes( mediaAttrib );
        }
        ValueAttribute* attrib = new ValueAttribute();
        attrib->setAttribute( "ptime" );
        LocalScopeAllocator lo;
        attrib->setValue( Data( UaConfiguration::instance()->getNetworkRtpRate() ).getData(lo) );

        //add the rtpmap attribute for the default codec
        SdpRtpMapAttribute* rtpMapAttrib = new SdpRtpMapAttribute();
        rtpMapAttrib->setPayloadType( 0 );
        rtpMapAttrib->setEncodingName( "PCMU" );
        rtpMapAttrib->setClockRate( 8000 );

        // add the value attribute just created to the media attribute object
        mediaAttrib->addValueAttribute( attrib );
        mediaAttrib->addmap( rtpMapAttrib );
        sipSdp->setSdpDescriptor( sdpDesc );

        //sipSdp->setRtpPort( atoi (UaConfiguration::instance()->getLocalRtpPort().c_str() )  );
        sipSdp->setRtpPort( UaDevice::instance()->getRtpPort() );

        call->setLocalSdp( new SipSdp( *sipSdp ) );
	int tmp;
        cpLog( LOG_DEBUG, "Local SDP:\n%s", call->getLocalSdp()->encodeBody(tmp).logData() );
    }

    // Save INVITE
    call->setRingInvite( new InviteMsg( inviteMsg ) );

    // Send INVITE
    sipEvent->getSipStack()->sendAsync( inviteMsg );

    Sptr < Contact > contact = new Contact( inviteMsg );
    // Set as current contact
    call->setContact( contact );
    // Add to contact list
    call->addContact( contact );

    return 0;
}


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