/// Remove identity info from a message.
void SipXauthIdentity::remove(SipMessage & message, HeaderName headerName)
{
int idHeaderCount = message.getCountHeaderFields(headerName);
if (idHeaderCount>0)
{
UtlString rUri;
message.getRequestUri(&rUri);
Os::Logger::instance().log(FAC_SIP, PRI_WARNING,
"SipXauthIdentity::remove"
": '%d' occurrances of %s in request to '%s'",
idHeaderCount, headerName, rUri.data());
for (int i = idHeaderCount - 1;i>=0;i--)
{
message.removeHeader(headerName, i);
}
}
}
/// Normalize identity info in a message.
void SipXauthIdentity::normalize(SipMessage & message, HeaderName headerName)
{
int idHeaderCount = message.getCountHeaderFields(headerName);
if (idHeaderCount>1)
{
UtlString rUri;
message.getRequestUri(&rUri);
Os::Logger::instance().log(FAC_SIP, PRI_WARNING,
"SipXauthIdentity::remove"
": '%d' occurrances of SipXauthIdentity in request to '%s'",
idHeaderCount, rUri.data());
// Remove all BUT the last header
for (int i = idHeaderCount - 2;i>=0;i--)
{
//message.removeHeader(SipXauthIdentity::AuthIdentityHeaderName, i);
message.removeHeader(headerName, i);
}
}
}
/// Check the signature and parse the identity contained in specified header name
bool SipXauthIdentity::decode(const UtlString& headerName,
const SipMessage& message,
const UtlString& callId,
const UtlString& fromTag,
DialogRule bindRule )
{
bool foundIdentityHeader = false;
UtlString rUri;
message.getRequestUri(&rUri);
int idHeaderCount = message.getCountHeaderFields(headerName);
if (1==idHeaderCount)
{
foundIdentityHeader =
decode(message.getHeaderValue(0, headerName),
callId, fromTag, bindRule);
}
else if (idHeaderCount>1)
{
Os::Logger::instance().log(FAC_SIP, PRI_DEBUG,
"SipXauthIdentity::decode:"
" '%d' occurrences of %s in request to '%s'",
idHeaderCount, headerName.data(), rUri.data());
foundIdentityHeader =
decode(message.getHeaderValue(idHeaderCount-1, headerName),
callId, fromTag, bindRule);
}
if (foundIdentityHeader)
{
Os::Logger::instance().log(FAC_SIP, PRI_DEBUG,
"SipXauthIdentity::decode:"
" found %s '%s' in request to '%s'",
headerName.data(), mIdentity.data(), rUri.data());
}
return foundIdentityHeader;
}
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);
}