uabase.cxx

SIP（Session Initiation Protocol）是由IETF定义

/* ====================================================================
 * The Vovida Software License, Version 1.0 
 * 
 * Copyright (c) 2000 Vovida Networks, Inc.  All rights reserved.
 * 
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 * modification, are permitted provided that the following conditions
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 * 1. Redistributions of source code must retain the above copyright
 *    notice, this list of conditions and the following disclaimer.
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 * 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].
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 * 4. Products derived from this software may not be called "VOCAL", nor
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 *    permission of Vovida Networks, Inc.
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 * ====================================================================
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 * 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
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 */


static const char* const UaBase_cxx_Version =
    "$Id: UaBase.cxx,v 1.12.2.3 2003/02/13 03:30:42 sprajpat Exp $";

#include "InviteMsg.hxx" 
#include "StatusMsg.hxx" 
#include "AckMsg.hxx" 
#include "SipRoute.hxx" 
#include "SystemInfo.hxx" 
#include "UaBase.hxx" 
#include "SipRecordRoute.hxx"
#include "SipVia.hxx"

using Vocal::InviteMsg;
using Vocal::StatusMsg;
using Vocal::UA::UaBase;

int Vocal::UA::UaBase::mySipPort = 5060;
NetworkAddress Vocal::UA::UaBase::myProxyAddress;
int Vocal::UA::UaBase::myTransport;
Data Vocal::UA::UaBase::myNAThost;

void
Vocal::UA::libSipInit(NetworkAddress& proxyAddess, int sipPort, Data NATHOST, int transport)
{
    Vocal::UA::UaBase::myProxyAddress = proxyAddess;
    if(Vocal::UA::UaBase::myProxyAddress.getPort() == -1)
    {
        Vocal::UA::UaBase::myProxyAddress.setPort(5060);
    }
    Vocal::UA::UaBase::mySipPort = sipPort;
    Vocal::UA::UaBase::myTransport = transport;
    Vocal::UA::UaBase::myNAThost =  NATHOST;
}

void 
UaBase::receivedMsg(const Sptr<SipMsg>& sipMsg)
{
    cpLog(LOG_DEBUG, "(%s) received message %s ", className().c_str(),
                      sipMsg->encode().logData());
    if(sipMsg->getType() == SIP_STATUS)
    {
        myState->recvStatus(*this, sipMsg);
    }
    else
    {
        myState->recvRequest(*this, sipMsg);
    }
}

void 
UaBase::sendMsg(const Sptr<SipMsg>& sipMsg)
{
    cpLog(LOG_DEBUG_STACK , "(%s) Processing message %s ", className().c_str(),
                      sipMsg->encode().logData());
    if(sipMsg->getType() == SIP_STATUS)
    {
        myState->sendStatus(*this, sipMsg);
    }
    else
    {
        myState->sendRequest(*this, sipMsg);
    }
}

Sptr<InviteMsg>
UaBase::createInvite(const Sptr<InviteMsg>& invMsg, bool changeCallId)
{
    Sptr<InviteMsg> inviteMsg = new InviteMsg(*invMsg);

    //Remove the contact header 
    inviteMsg->setNumContact(0);
    //Set the contact as B2b
    Sptr<BaseUrl> bUrl = inviteMsg->getFrom().getUrl();
    Sptr<SipUrl> mUrl;
    mUrl.dynamicCast(bUrl);
    //TODO - Handle the case with TelUrl
    assert(mUrl != 0);
   
    //mUrl->setHost(theSystem.gethostAddress());
    mUrl->setHost((UaBase::myNAThost));
    mUrl->setPort(Data(mySipPort));
 
    SipContact me;
    me.setUrl(mUrl);
    inviteMsg->setContact(me);

    //Clear VIA and set B2b as first via
    inviteMsg->flushViaList();
    SipVia via;
    via.setHost(theSystem.gethostAddress());
    via.setPort(mySipPort);
    if(myTransport == 2) //TCP
    {
        via.setTransport("TCP");
    }
    inviteMsg->setVia(via);

    inviteMsg->setNumRecordRoute(0);
    inviteMsg->setNumRoute(0);

    if(changeCallId)
    {
        //Set the new callId 
        SipCallId id;
        inviteMsg->setCallId(id);
        //Set the request URI to outbound proxy
        SipRequestLine& reqLine = inviteMsg->getMutableRequestLine();
        Sptr<SipUrl> sUrl;
        sUrl.dynamicCast(reqLine.getUrl());
        sUrl->setMaddrParam("");
        //Look for magic "%" in the request line
        //if found parse it and send it to the contact found in it
        string sUser = sUrl->getUserValue().logData();
        string::size_type pos;
        if((pos = sUser.find("%")) != string::npos)
        {
            string user = sUser.substr(0, pos);
            string hostPort = sUser.substr(pos+1);
            NetworkAddress na (hostPort);
            string nextHopAddr = na.getIpName().convertString();
            int nextHopPort = na.getPort();
            if(nextHopPort == -1) nextHopPort = 5060;
            sUrl->setHost(nextHopAddr);
            sUrl->setPort(Data(nextHopPort));
 
        }
        else
        {
            string outboundProxyAddr = myProxyAddress.getIpName().convertString();
            int outboundProxyPort =  myProxyAddress.getPort();
            sUrl->setHost(outboundProxyAddr);
            sUrl->setPort(Data(outboundProxyPort));
            if(!sUrl->getUserValue().length())
            {
               //set the user to be to
               sUrl->setUserValue(inviteMsg->getTo().getUser());
            }
        }
        if(myTransport == 2) //TCP
        {
            sUrl->setTransportParam("tcp");
        }
        reqLine.setUrl(sUrl);
    }

    return inviteMsg;
}

Sptr<InviteMsg>
UaBase::createReInvite(const Sptr<InviteMsg>& invMsg)
{
    Sptr<InviteMsg> inviteMsg = createInvite(invMsg, false);

    //This is a reInvite case, will use the route we have saved in the
    //begining of the dialog

    assert(myRouteList.size());
    inviteMsg->setRouteList(myRouteList);
    SipRoute siproute = inviteMsg->getRoute(0);
    inviteMsg->removeRoute(0);
    SipRequestLine& reqLine = inviteMsg->getMutableRequestLine();
    reqLine.setUrl( siproute.getUrl() );
    inviteMsg->setCallId(getRequest()->getCallId());
    //Get the CSeq from the local CSeq, increment it and use it.
    myLocalCSeq.incrCSeq();
    inviteMsg->setCSeq(myLocalCSeq);
    return inviteMsg;
}

      
//Sptr<StatusMsg> // VEER
void
UaBase::sendReplyForRequest(const Sptr<SipMsg>& sipMsg, int statusCode)
{
    assert(sipMsg->getType() != SIP_STATUS);
    //Send status message
    Sptr<SipCommand> sipCmd;
    sipCmd.dynamicCast(sipMsg);
    assert(sipCmd != 0);
    Sptr<StatusMsg> sendSMsg = new StatusMsg(*sipCmd, statusCode);
    cpLog(LOG_DEBUG, "(%s) sending status %s", className().c_str(), sendSMsg->encode().logData());
    myStack->sendReply(sendSMsg);
    setResponse(sendSMsg);
    // return sendSMsg; // VEER
}

void
UaBase::ackStatus(const Sptr<SipMsg>& msg)
{
    Sptr<StatusMsg> sMsg;
    sMsg.dynamicCast(msg);
    assert(sMsg != 0);
    if(sMsg->getStatusLine().getStatusCode() >= 200)
    {
        Sptr<AckMsg> ackMsg = new AckMsg(*sMsg);
        //if(sMsg->getStatusLine().getStatusCode() > 200)
        {
            //Ack should have the same request line and via as the
            //Original invite
            Sptr<SipCommand> invMsg;
            invMsg.dynamicCast(getRequest());
            assert(invMsg != 0);
            ackMsg->flushViaList();
 
            SipVia via = invMsg->getVia(0);
            ackMsg->setVia(via, 0);
            ackMsg->setRouteList(myRouteList);
            ackMsg->removeRoute(0);
 
            Sptr<SipUrl> sUrl;
            sUrl.dynamicCast(invMsg->getRequestLine().getUrl());
            SipRequestLine rLine(SIP_ACK, sUrl);
            ackMsg->setRequestLine(rLine);
        }
        myStack->sendAsync(ackMsg);
    }
}



UaBase::~UaBase()
{
    //cerr << "UaBase::~UaBase:" << myAgentRole << endl;
    clearRouteList();
}

void
UaBase::clearRouteList()
{
    //cleanup the routelist
    while(myRouteList.size())
    {
        SipRoute* r = myRouteList.back();
        myRouteList.pop_back();
        delete r;
    }
}


void 
UaBase::saveRouteList(const Sptr<SipMsg>& msg, bool reverse)
{
    clearRouteList();
    SipRecordRouteList rrList = msg->getrecordrouteList();
    SipRecordRouteList::iterator iter = rrList.begin();

    //Reverse the recordRoute list and form a route list
    // (A,B,C) -> (C,B, A)
    while ( iter != rrList.end() )
    {
        Sptr< BaseUrl > baseUrl = (*iter)->getUrl();
        if( baseUrl->getType() == TEL_URL )
        {
            cpLog( LOG_ERR, "TEL_URL currently not supported\n" );
            continue;
        }
        Sptr< SipUrl > sipUrl;
        sipUrl.dynamicCast( baseUrl );
        assert( sipUrl != 0 );

        SipRoute* route = new SipRoute;
cpLog(LOG_DEBUG, "**** Adding route %d: %s", reverse, baseUrl->encode().logData());
        if(reverse)
        {
            route->setUrl( baseUrl );
            myRouteList.insert( myRouteList.begin(), route );
        }
        else
        {
            route->setUrl( baseUrl );
            myRouteList.push_back( route );
        }
        iter++;
    }

    //Append the contacts at the end of the route list
    int numContact = msg->getNumContact();
    if ( numContact )
    {
        SipContact contact = msg->getContact( numContact - 1 );
        SipRoute* route = new SipRoute;
        route->setUrl( contact.getUrl() );
        myRouteList.push_back( route );
    }
}