UtlBoolean SipRefreshManager::getAcceptedExpiration(RefreshDialogState* state,
const SipMessage& sipResponse,
int& expirationPeriod)
{
UtlString method;
UtlBoolean foundExpiration = FALSE;
int cseq;
sipResponse.getCSeqField(&cseq, &method);
if(method.compareTo(SIP_REGISTER_METHOD) == 0)
{
// Get the presumably first contact in the REGISTER request
// so that we can find the same contact in the response and
// find out what the expiration is
UtlString requestContact;
Url requestContactUri;
if(state && state->mpLastRequest &&
state->mpLastRequest->getContactEntry(0, &requestContact))
{
requestContactUri = requestContact;
}
// Register could have it in the Contact header
UtlString responseContactValue;
int contactIndex = 0;
while(sipResponse.getContactEntry(contactIndex , &responseContactValue))
{
// Get the expires parameter for the contact if it exists
Url contactUri(responseContactValue);
if(requestContactUri.isUserHostPortEqual(contactUri))
{
UtlString contactExpiresParameter;
if(contactUri.getFieldParameter(SIP_EXPIRES_FIELD,
contactExpiresParameter) &&
!contactExpiresParameter.isNull())
{
foundExpiration = TRUE;
// Convert to int
expirationPeriod = atoi(contactExpiresParameter);
}
}
contactIndex++;
}
}
if(!foundExpiration)
{
// Not sure if we care if this is a request or response
foundExpiration = sipResponse.getExpiresField(&expirationPeriod);
}
return(foundExpiration);
}
UtlBoolean SipRefreshManager::getInitialExpiration(const SipMessage& sipRequest,
int& expirationPeriod)
{
UtlString method;
UtlBoolean foundExpiration = FALSE;
sipRequest.getRequestMethod(&method);
if(method.compareTo(SIP_REGISTER_METHOD) == 0)
{
// Register could have it in the Contact header
UtlString requestContactValue;
if(sipRequest.getContactEntry(0 , &requestContactValue))
{
// Get the expires parameter for the contact if it exists
Url contactUri(requestContactValue);
UtlString contactExpiresParameter;
if(contactUri.getFieldParameter(SIP_EXPIRES_FIELD,
contactExpiresParameter) &&
!contactExpiresParameter.isNull())
{
foundExpiration = TRUE;
// Convert to int
expirationPeriod = atoi(contactExpiresParameter);
}
}
}
if(!foundExpiration)
{
// Not sure if we care if this is a request or response
foundExpiration = sipRequest.getExpiresField(&expirationPeriod);
}
return(foundExpiration);
}
AuthPlugin::AuthResult
TransferControl::authorizeAndModify(const UtlString& id, /**< The authenticated identity of the
* request originator, if any (the null
* string if not).
* This is in the form of a SIP uri
* identity value as used in the
* credentials database (user@domain)
* without the scheme or any parameters.
*/
const Url& requestUri, ///< parsed target Uri
RouteState& routeState, ///< the state for this request.
const UtlString& method,///< the request method
AuthResult priorResult,///< results from earlier plugins.
SipMessage& request, ///< see AuthPlugin wrt modifying
bool bSpiralingRequest, ///< request spiraling indication
UtlString& reason ///< rejection reason
)
{
AuthResult result = CONTINUE;
// get the call-id to use in logging
UtlString callId;
request.getCallIdField(&callId);
UtlString hostAddress;
int hostPort;
UtlString hostProtocols;
//requestUri.getHostAddress(hostAddress);
//request.getContactUri(0, &hostAddress);;
request.getContactAddress(0, &hostAddress,&hostPort,&hostProtocols);
if (DENY != priorResult)
{
if (method.compareTo(SIP_REFER_METHOD) == 0)
{
UtlString targetStr;
if (request.getReferToField(targetStr))
{
Url target(targetStr, Url::NameAddr); // parse the target URL
UtlString targetMethod;
if ( Url::SipUrlScheme == target.getScheme()
/* REFER can create requests other than INVITE: we don't care about those *
* so check that the method is INVITE or is unspecified (INVITE is the default) */
&& ( ! target.getUrlParameter(SIP_METHOD_URI_PARAMETER, targetMethod)
|| (0==targetMethod.compareTo(SIP_INVITE_METHOD, UtlString::ignoreCase))
))
{
if (id.isNull())
{
// UnAuthenticated REFER. Do challenge the REFER to confirm the
// identity of the transferor. Note: prior to XECS-2487, we used to challenge
// only the unauthenticated REFERs that didn't carry a Replaces header.
// The fix for XECS-2487 now requires that all unauthenticated REFERs
// be challenged so that consultative transfers get routed properly
// when user-based gateway section is used. See tracker for the details
if (mpSipRouter->isLocalDomain(target))
{
//White list of two servets to let Exchange REFER to sipXecs endpoints
if (hostAddress.compareTo(server1, UtlString::ignoreCase) == 0 || hostAddress.compareTo(server2, UtlString::ignoreCase) == 0)
{
Os::Logger::instance().log(FAC_AUTH, PRI_INFO, "TransferControl[%s]::authorizeAndModify "
"Whitelist host '%s' in call '%s'",
mInstanceName.data(),hostAddress.data(),callId.data()
);
result = ALLOW; //Whitelist matched so allow the transfer
}else{
Os::Logger::instance().log(FAC_AUTH, PRI_INFO, "TransferControl[%s]::authorizeAndModify "
"challenging transfer in call '%s' from host '%s'",
mInstanceName.data(), callId.data(),hostAddress.data()
);
result = DENY; // we need an identity to attach to the Refer-To URI
}
}
else
{
/*
* This is a transfer to a target outside our domain, so let it go
* unchallenged. See XECS-806
*/
Os::Logger::instance().log(FAC_AUTH, PRI_DEBUG, "TransferControl[%s]::authorizeAndModify "
"allowing foriegn transfer in call '%s'",
mInstanceName.data(), callId.data()
);
// Add the References to the refer-to. Adding the callId field as a reference
// header (will be used in resulting INVITE) in the Refer-To provides us
// with enough information to be able to logically tie the calls together.
// Useful for CDR records.
UtlString refcallId(callId);
refcallId.append(";rel=refer");
target.setHeaderParameter(SIP_REFERENCES_FIELD, refcallId.data());
Os::Logger::instance().log(FAC_AUTH, PRI_DEBUG, "TransferControl[%s]::authorizeAndModify "
"adding Reference field [%s] to refer-to",
mInstanceName.data(), callId.data()
);
request.setReferToField(target.toString().data());
result = ALLOW;
}
}
else
{
UtlString contactString;
request.getContactEntry(0, &contactString);
Url contactUri( contactString );
UtlString userId;
contactUri.getUserId(contactString);
Os::Logger::instance().log(FAC_AUTH, PRI_DEBUG, "TransferControl::authorizeAndModify - Contact field is: %s ", contactString.data());
if (contactString != "callcontroller") {
// Authenticated REFER
// annotate the refer-to with the authenticated controller identity
SipXauthIdentity controllerIdentity;
controllerIdentity.setIdentity(id);
controllerIdentity.encodeUri(target);
// add the References to the refer-to.
UtlString refcallId(callId);
refcallId.append(";rel=refer");
target.setHeaderParameter(SIP_REFERENCES_FIELD, refcallId.data());
Os::Logger::instance().log(FAC_AUTH, PRI_DEBUG, "TransferControl[%s]::authorizeAndModify "
"adding Reference field [%s] to refer-to",
mInstanceName.data(), callId.data()
);
request.setReferToField(target.toString().data());
}
}
}
else
{
Os::Logger::instance().log(FAC_AUTH, PRI_WARNING, "TransferControl[%s]::authorizeAndModify "
"unrecognized refer target '%s' for call '%s'",
mInstanceName.data(), targetStr.data(), callId.data()
);
}
}
else
{
// REFER without a Refer-To header... incorrect, but just ignore it.
Os::Logger::instance().log(FAC_AUTH, PRI_WARNING,
"TransferControl[%s]::authorizeAndModify "
"REFER method without Refer-To in call '%s'",
mInstanceName.data(), callId.data()
);
}
}
else if (method.compareTo(SIP_INVITE_METHOD) == 0)
{
UtlString targetCallId;
UtlString targetFromTag;
UtlString targetToTag;
if (request.getReplacesData(targetCallId, targetToTag, targetFromTag))
{
/*
* This is an INVITE with Replaces: probably either the completion
* of a call pickup or a consultative transfer.
* In any case, it will not create a new call - just connect something
* to an existing call - so we don't need to make any new authorization
* decisions.
*/
result = ALLOW;
}
else
{
// INVITE without Replaces: is not a transfer - ignore it.
}
}
else
{
// neither REFER nor INVITE, so is not a transfer - ignore it.
}
}
else
{
// Some earlier plugin already denied this - don't waste time figuring it out.
Os::Logger::instance().log(FAC_AUTH, PRI_DEBUG, "TransferControl[%s]::authorizeAndModify "
"prior authorization result %s for call %s",
mInstanceName.data(), AuthResultStr(priorResult), callId.data()
);
}
return result;
}
UtlBoolean SipPersistentSubscriptionMgr::updateDialogInfo(
const SipMessage& subscribeRequest,
UtlString& resourceId,
UtlString& eventTypeKey,
UtlString& eventType,
UtlString& subscribeDialogHandle,
UtlBoolean& isNew,
UtlBoolean& isSubscriptionExpired,
SipMessage& subscribeResponse,
SipSubscribeServerEventHandler& handler)
{
OsSysLog::add(FAC_SIP, PRI_DEBUG,
"SipPersistentSubscriptionMgr::updateDialogInfo "
"resourceId = '%s', eventTypeKey = '%s'",
resourceId.data(), eventTypeKey.data());
UtlBoolean ret;
// Call SipSubscriptionMgr to update the in-memory data.
ret = SipSubscriptionMgr::updateDialogInfo(subscribeRequest,
resourceId,
eventTypeKey,
eventType,
subscribeDialogHandle,
isNew,
isSubscriptionExpired,
subscribeResponse,
handler);
// If that succeeded, update the IMDB.
if (ret)
{
UtlString requestUri;
UtlString callId;
UtlString contactEntry;
UtlString to;
UtlString from;
UtlString route;
UtlString accept;
subscribeRequest.getRequestUri(&requestUri);
subscribeRequest.getCallIdField(&callId);
subscribeRequest.getContactEntry(0, &contactEntry);
subscribeRequest.getToField(&to);
subscribeRequest.getFromField(&from);
subscribeRequest.buildRouteField(&route);
accept.append(subscribeRequest.getHeaderValue(0, SIP_ACCEPT_FIELD));
int expires = 0;
subscribeResponse.getExpiresField(&expires);
expires += OsDateTime::getSecsSinceEpoch();
int subscribeCseq;
UtlString subscribeCseqMethod;
subscribeRequest.getCSeqField(&subscribeCseq, &subscribeCseqMethod);
OsSysLog::add(FAC_SIP, PRI_DEBUG,
"SipPersistentSubscriptionMgr::updateDialogInfo "
"mComponent = '%s', requestUri = '%s', callId = '%s', contactEntry = '%s', expires = %d, to = '%s', from = '%s', key = '%s', route = '%s', accept = '%s'",
mComponent.data(), requestUri.data(),
callId.data(), contactEntry.data(), expires,
to.data(), from.data(), resourceId.data(), route.data(),
accept.data());
// Attempt to update an existing row.
int now = (int)OsDateTime::getSecsSinceEpoch();
ret = mSubscriptionDBInstance->updateSubscribeUnexpiredSubscription(
mComponent, to, from, callId, eventTypeKey, "",
now, expires, subscribeCseq);
if (!ret)
{
// Add a new row.
// This call assumes that eventTypeKey is OK for use as the <eventtype>,
// and that the NOTIFY CSeq's will start at 1. 0 is used as
// the initial XML version.
ret = mSubscriptionDBInstance->insertRow(
mComponent, requestUri, callId, contactEntry,
expires, subscribeCseq, eventTypeKey, "",
to, from, resourceId, route, 1, accept, 0);
if (!ret)
{
OsSysLog::add(FAC_SIP, PRI_ERR,
"SipPersistantSubscriptionMgr::addSubscription "
"Could not update or insert record in database");
}
}
// Start the save timer.
mPersistenceTimer.oneshotAfter(sPersistInterval);
}
return ret;
}
UtlBoolean SipXProxyCseObserver::handleMessage(OsMsg& eventMessage)
{
int msgType = eventMessage.getMsgType();
switch (msgType)
{
case OsMsg::OS_EVENT:
switch (eventMessage.getMsgSubType())
{
case OsEventMsg::NOTIFY:
if (mpWriter)
{
mpWriter->flush();
}
break;
}
break ;
case OsMsg::PHONE_APP:
{
SipMessage* sipMsg;
if(SipMessageEvent::TRANSPORT_ERROR == ((SipMessageEvent&)eventMessage).getMessageStatus())
{
OsSysLog::add(FAC_SIP, PRI_ERR,
"SipXProxyCseObserver::handleMessage transport error");
}
else if((sipMsg = (SipMessage*)((SipMessageEvent&)eventMessage).getMessage()))
{
UtlString method;
int rspStatus = 0;
UtlString rspText;
UtlString contact;
UtlString toTag;
enum
{
UnInteresting,
aCallRequest,
aCallSetup,
aCallFailure,
aCallEnd,
aCallTransfer
} thisMsgIs = UnInteresting;
Url toUrl;
sipMsg->getToUrl(toUrl);
// explicitly, an INVITE Request
toUrl.getFieldParameter("tag", toTag);
if (!sipMsg->isResponse())
{
// sipMsg is a Request
sipMsg->getRequestMethod(&method);
if (0==method.compareTo(SIP_INVITE_METHOD, UtlString::ignoreCase))
{
if (toTag.isNull())
{
thisMsgIs = aCallRequest;
}
}
else if (0==method.compareTo(SIP_REFER_METHOD, UtlString::ignoreCase))
{
thisMsgIs = aCallTransfer;
sipMsg->getContactEntry(0, &contact);
}
else if (0==method.compareTo(SIP_BYE_METHOD, UtlString::ignoreCase))
{
thisMsgIs = aCallEnd; // no additional information needed
}
else
{
// other request methods are not interesting
}
}
else // this is a response
{
int seq;
if (sipMsg->getCSeqField(&seq, &method)) // get the method out of cseq field
{
if (0==method.compareTo(SIP_INVITE_METHOD, UtlString::ignoreCase))
{
// Responses to INVITES are handled differently based on whether
// or not the INVITE is dialog-forming. If dialog-forming,
// any final response above 400 is considered a failure for CDR
// purposes. If not dialog-forming, then any final response above 400
// except 401 Unauthorized, 407 Proxy Authentication Required and
// 408 Request Timeout will terminate. If we're in a dialog then
// only 408 (Request Timeout) and 481 (Call/Transaction does not exist)
// will terminate the dialog.
rspStatus = sipMsg->getResponseStatusCode();
if (rspStatus >= SIP_4XX_CLASS_CODE) // any failure
{
// a failure code - this is a potential CallFailure - Call Resolver will determine.
thisMsgIs = aCallFailure;
sipMsg->getResponseStatusText(&rspText);
}
else if ( ( rspStatus >= SIP_2XX_CLASS_CODE )
&& ( rspStatus < SIP_3XX_CLASS_CODE )
)
{
thisMsgIs = aCallSetup;
sipMsg->getContactEntry(0, &contact);
}
}
else
{
// responses to non-INVITES are not interesting
}
}
else
{
OsSysLog::add(FAC_SIP, PRI_ERR, "SipXProxyCseObserver - no Cseq in response");
}
}
# ifdef LOG_DEBUG
OsSysLog::add(FAC_SIP, PRI_DEBUG, "SipXProxyCseObserver message is %s",
( thisMsgIs == UnInteresting ? "UnInteresting"
: thisMsgIs == aCallEnd ? "a Call End"
: thisMsgIs == aCallFailure ? "a Call Failure"
: thisMsgIs == aCallRequest ? "a call Request"
: thisMsgIs == aCallSetup ? "a Call Setup"
: thisMsgIs == aCallTransfer ? "a Call Transfer"
: "BROKEN"
));
# endif
if (thisMsgIs != UnInteresting)
{
// collect the sequence data
mSequenceNumber++;
OsTime timeNow;
OsDateTime::getCurTime(timeNow);
// collect the dialog information
UtlString callId;
sipMsg->getCallIdField(&callId);
Url toUrl;
sipMsg->getToUrl(toUrl);
UtlString toTag;
toUrl.getFieldParameter("tag", toTag);
Url fromUrl;
sipMsg->getFromUrl(fromUrl);
UtlString fromTag;
fromUrl.getFieldParameter("tag", fromTag);
// collect the To and From
UtlString toField;
sipMsg->getToField(&toField);
UtlString fromField;
sipMsg->getFromField(&fromField);
UtlString referTo;
UtlString referredBy;
UtlString requestUri;
sipMsg->getReferToField(referTo);
sipMsg->getReferredByField(referredBy);
sipMsg->getRequestUri(&requestUri);
UtlString responseMethod;
int cseqNumber;
sipMsg->getCSeqField(&cseqNumber, &responseMethod);
// generate the call state event record
if (mpBuilder)
{
switch (thisMsgIs)
{
case aCallRequest:
mpBuilder->callRequestEvent(mSequenceNumber, timeNow, contact);
break;
case aCallSetup:
mpBuilder->callSetupEvent(mSequenceNumber, timeNow, contact);
break;
case aCallFailure:
mpBuilder->callFailureEvent(mSequenceNumber, timeNow, rspStatus, rspText);
break;
case aCallEnd:
mpBuilder->callEndEvent(mSequenceNumber, timeNow);
break;
case aCallTransfer:
mpBuilder->callTransferEvent(mSequenceNumber, timeNow,
contact, referTo, referredBy, requestUri);
break;
default:
// shouldn't be possible to get here
OsSysLog::add(FAC_SIP, PRI_ERR, "SipXProxyCseObserver invalid thisMsgIs");
break;
}
mpBuilder->addCallData(cseqNumber, callId, fromTag, toTag, fromField, toField);
UtlString via;
for (int i=0; sipMsg->getViaField(&via, i); i++)
{
mpBuilder->addEventVia(via);
}
mpBuilder->completeCallEvent();
// get the completed record
UtlString event;
mpBuilder->finishElement(event);
if (mpWriter)
{
mpWriter->writeLog(event.data());
}
}
else
{
OsSysLog::add(FAC_SIP, PRI_ERR, "SipXProxyCseObserver - no CallStateEventBuilder!");
}
}
}
else
{
OsSysLog::add(FAC_SIP, PRI_ERR, "SipXProxyCseObserver getMessage returned NULL");
}
}
break;
default:
{
OsSysLog::add(FAC_SIP, PRI_ERR, "SipXProxyCseObserver invalid message type %d", msgType );
}
} // end switch (msgType)
return(TRUE);
}
UtlBoolean SipXProxyCseObserver::handleMessage(OsMsg& eventMessage)
{
int msgType = eventMessage.getMsgType();
switch (msgType)
{
case OsMsg::OS_EVENT:
switch (eventMessage.getMsgSubType())
{
case OsEventMsg::NOTIFY:
if (mpWriter)
{
mpWriter->flush();
}
break;
}
break ;
case OsMsg::PHONE_APP:
{
SipMessage* sipMsg;
if(SipMessageEvent::TRANSPORT_ERROR == ((SipMessageEvent&)eventMessage).getMessageStatus())
{
Os::Logger::instance().log(FAC_SIP, PRI_ERR,
"SipXProxyCseObserver::handleMessage transport error");
}
else if((sipMsg = (SipMessage*)((SipMessageEvent&)eventMessage).getMessage()))
{
UtlString method;
int rspStatus = 0;
UtlString rspText;
UtlString contact;
UtlString toTag;
enum
{
UnInteresting,
aCallRequest,
aCallSetup,
aCallFailure,
aCallEnd,
aCallTransfer
} thisMsgIs = UnInteresting;
Url toUrl;
sipMsg->getToUrl(toUrl);
// explicitly, an INVITE Request
toUrl.getFieldParameter("tag", toTag);
if (!sipMsg->isResponse())
{
// sipMsg is a Request
sipMsg->getRequestMethod(&method);
if (0==method.compareTo(SIP_INVITE_METHOD, UtlString::ignoreCase))
{
if (toTag.isNull())
{
sipMsg->getContactEntry(0, &contact);
thisMsgIs = aCallRequest;
}
}
else if (0==method.compareTo(SIP_REFER_METHOD, UtlString::ignoreCase))
{
thisMsgIs = aCallTransfer;
sipMsg->getContactEntry(0, &contact);
}
else if (0==method.compareTo(SIP_BYE_METHOD, UtlString::ignoreCase))
{
thisMsgIs = aCallEnd; // no additional information needed
}
else
{
// other request methods are not interesting
}
}
else // this is a response
{
int seq;
if (sipMsg->getCSeqField(&seq, &method)) // get the method out of cseq field
{
if (0==method.compareTo(SIP_INVITE_METHOD, UtlString::ignoreCase))
{
// Responses to INVITES are handled differently based on whether
// or not the INVITE is dialog-forming. If dialog-forming,
// any final response above 400 is considered a failure for CDR
// purposes. If not dialog-forming, then any final response above 400
// except 401 Unauthorized, 407 Proxy Authentication Required and
// 408 Request Timeout will terminate. If we're in a dialog then
// only 408 (Request Timeout) and 481 (Call/Transaction does not exist)
// will terminate the dialog.
rspStatus = sipMsg->getResponseStatusCode();
if (rspStatus >= SIP_4XX_CLASS_CODE) // any failure
{
// a failure code - this is a potential CallFailure - Call Resolver will determine.
thisMsgIs = aCallFailure;
sipMsg->getResponseStatusText(&rspText);
}
else if ( ( rspStatus >= SIP_2XX_CLASS_CODE )
&& ( rspStatus < SIP_3XX_CLASS_CODE )
)
{
thisMsgIs = aCallSetup;
sipMsg->getContactEntry(0, &contact);
}
}
else
{
// responses to non-INVITES are not interesting
}
}
else
{
Os::Logger::instance().log(FAC_SIP, PRI_ERR, "SipXProxyCseObserver - no Cseq in response");
}
}
# ifdef LOG_DEBUG
Os::Logger::instance().log(FAC_SIP, PRI_DEBUG, "SipXProxyCseObserver message is %s",
( thisMsgIs == UnInteresting ? "UnInteresting"
: thisMsgIs == aCallEnd ? "a Call End"
: thisMsgIs == aCallFailure ? "a Call Failure"
: thisMsgIs == aCallRequest ? "a call Request"
: thisMsgIs == aCallSetup ? "a Call Setup"
: thisMsgIs == aCallTransfer ? "a Call Transfer"
: "BROKEN"
));
# endif
if (thisMsgIs != UnInteresting)
{
// collect the sequence data
mSequenceNumber++;
OsTime timeNow;
OsDateTime::getCurTime(timeNow);
// collect the dialog information
UtlString callId;
sipMsg->getCallIdField(&callId);
Url toUrl;
sipMsg->getToUrl(toUrl);
UtlString toTag;
toUrl.getFieldParameter("tag", toTag);
Url fromUrl;
sipMsg->getFromUrl(fromUrl);
UtlString fromTag;
fromUrl.getFieldParameter("tag", fromTag);
// collect the To and From
UtlString toField;
sipMsg->getToField(&toField);
UtlString fromField;
sipMsg->getFromField(&fromField);
// collect the branch Id (i.e. transaction id) and via count.
UtlString viaValue;
int viaCount;
UtlString branchId;
viaCount = sipMsg->getCountHeaderFields(SIP_VIA_FIELD);
viaCount = viaCount + sipMsg->getCountHeaderFields(SIP_SHORT_VIA_FIELD);
if ( sipMsg->getViaFieldSubField( &viaValue, 0 ) ) {
sipMsg->getViaTag( viaValue, "branch", branchId );
}
UtlString referTo;
UtlString referredBy;
UtlString requestUri;
UtlString references;
UtlString replaces_callId;
UtlString replaces_toTag;
UtlString replaces_fromTag;
UtlString matchingIdentityHeader;
SipXauthIdentity sipxIdentity(*sipMsg, matchingIdentityHeader, true,SipXauthIdentity::allowUnbound);
sipMsg->getReferToField(referTo);
sipMsg->getReferredByField(referredBy);
sipMsg->getRequestUri(&requestUri);
sipMsg->getReferencesField(&references);
if (sipMsg->getReplacesData(replaces_callId, replaces_toTag, replaces_fromTag)) {
if (references.length() != 0) {
references.append(",");
}
references.append(replaces_callId);
references.append(";rel=xfer");
}
UtlString responseMethod;
UtlString calleeRoute;
int cseqNumber;
sipMsg->getCSeqField(&cseqNumber, &responseMethod);
BranchTimePair* callIdBranchIdTime;
// generate the call state event record
if (mpBuilder)
{
UtlString identity;
UtlString recordRoute;
bool routeFound = false;
bool paiPresent = false;
switch (thisMsgIs)
{
case aCallRequest:
if (sipxIdentity.getIdentity(identity)) {
paiPresent = true;
}
if ( branchId && branchId.data() ) {
mCallTransMutex.acquire();
unsigned long currentTime = OsDateTime::getSecsSinceEpoch();
UtlString* keyCallId = new UtlString(callId);
BranchTimePair* valBranchTimePair = new BranchTimePair(branchId.data(), ¤tTime, &paiPresent);
if (NULL == mCallTransMap.insertKeyAndValue(keyCallId, valBranchTimePair) ) {
// Unable to add callId to map so it must already be present.
delete keyCallId;
delete valBranchTimePair;
// Check if the paiPresent value is set to true or not.
// If not set and we now have a PAI for this call, set it and generate another call request state event
// with this info. Otherwise skip over.
if ( paiPresent ) {
callIdBranchIdTime = (BranchTimePair*) mCallTransMap.findValue(&callId);
if ( callIdBranchIdTime && (*callIdBranchIdTime->getPaiPresent() == false) ) {
// need to generate another call request event in order to state originator is internal.
callIdBranchIdTime->setPaiPresent(&paiPresent);
}
else {
mCallTransMutex.release();
return(TRUE);
}
}
else {
mCallTransMutex.release();
return(TRUE);
}
}
mCallTransMutex.release();
}
mpBuilder->callRequestEvent(mSequenceNumber, timeNow, contact, references, branchId, viaCount, paiPresent);
break;
case aCallSetup:
// Clear out from the map only if rspStatus is higher than 200 as its possible to receive multiple 200 messages.
// If the response is 200, the call in the map will be cleared out when the call ends.
mCallTransMutex.acquire();
callIdBranchIdTime = (BranchTimePair*) mCallTransMap.findValue(&callId);
if ( callIdBranchIdTime && (0 == branchId.compareTo(callIdBranchIdTime)) ) {
if ( rspStatus > SIP_2XX_CLASS_CODE ) {
mCallTransMap.destroy(&callId);
}
mCallTransMutex.release();
}
else
{
// CallId/BranchId are either not found or doesn't match. Not a final response.
mCallTransMutex.release();
return(TRUE);
}
for (int rrNum = 0;
(!routeFound && sipMsg->getRecordRouteUri(rrNum, &recordRoute));
rrNum++
)
{
Url recordRouteUrl(recordRoute);
if (mpSipUserAgent->isMyHostAlias(recordRouteUrl)) {
// This is a record route for our proxy, extract Call tags if they exist.
recordRouteUrl.getUrlParameter(SIP_SIPX_CALL_DEST_FIELD, calleeRoute, 0);
routeFound = true;
}
}
mpBuilder->callSetupEvent(mSequenceNumber, timeNow, contact, calleeRoute, branchId, viaCount);
break;
case aCallFailure:
// Failure case means that the response code is > 400. If the call is found
// in the map, then this is a final response. Delete from the map and build an event.
mCallTransMutex.acquire();
callIdBranchIdTime = (BranchTimePair*) mCallTransMap.findValue(&callId);
if ( callIdBranchIdTime && (0 == branchId.compareTo(callIdBranchIdTime)) ) {
mCallTransMap.destroy(&callId);
mCallTransMutex.release();
if ( rspStatus != SIP_PROXY_AUTH_REQ_CODE ) {
mpBuilder->callFailureEvent(mSequenceNumber, timeNow, branchId, viaCount, rspStatus, rspText);
}
else {
// response was an authentication required. Don't build a CSE for these as a new Invite will
// occur.
return(TRUE);
}
}
else
{
// Call was not found in the map so this is not a final response. Ignore it.
mCallTransMutex.release();
return(TRUE);
}
break;
case aCallEnd:
mCallTransMutex.acquire();
mCallTransMap.destroy(&callId);
mCallTransMutex.release();
mpBuilder->callEndEvent(mSequenceNumber, timeNow);
break;
case aCallTransfer:
mpBuilder->callTransferEvent(mSequenceNumber, timeNow,
contact, referTo, referredBy, requestUri);
break;
default:
// shouldn't be possible to get here
Os::Logger::instance().log(FAC_SIP, PRI_ERR, "SipXProxyCseObserver invalid thisMsgIs");
break;
}
mpBuilder->addCallData(cseqNumber, callId, fromTag, toTag, fromField, toField);
UtlString via;
for (int i=0; sipMsg->getViaField(&via, i); i++)
{
mpBuilder->addEventVia(via);
}
mpBuilder->completeCallEvent();
// get the completed record
UtlString event;
mpBuilder->finishElement(event);
if (mpWriter)
{
mpWriter->writeLog(event.data());
}
}
else
{
Os::Logger::instance().log(FAC_SIP, PRI_ERR, "SipXProxyCseObserver - no CallStateEventBuilder!");
}
}
}
else
{
Os::Logger::instance().log(FAC_SIP, PRI_ERR, "SipXProxyCseObserver getMessage returned NULL");
}
}
break;
default:
{
Os::Logger::instance().log(FAC_SIP, PRI_ERR, "SipXProxyCseObserver invalid message type %d", msgType );
}
} // end switch (msgType)
return(TRUE);
}