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openbts/SIP/SIPDialog.cpp

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/*
* Copyright 2013, 2014 Range Networks, Inc.
*
* This software is distributed under multiple licenses;
* see the COPYING file in the main directory for licensing
* information for this specific distribution.
*
* This use of this software may be subject to additional restrictions.
* See the LEGAL file in the main directory for details.
This program is distributed in the hope that it will be useful,
but WITHOUT ANY WARRANTY; without even the implied warranty of
MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.
*/
// Written by Pat Thompson.
#define LOG_GROUP LogGroup::SIP // Can set Log.Level.SIP for debugging
#include "SIPBase.h"
#include "SIPDialog.h"
#include "SIP2Interface.h"
#include "SIPTransaction.h"
#include <Reporting.h> // For gReports.
#include <L3TranEntry.h>
#include <L3TermCause.h>
#include <L3StateMachine.h>
#include <GSML3MMElements.h> // for L3CMServiceType
#include <GSML3CCElements.h> // for L3Cause
#include <algorithm>
namespace SIP {
using namespace Control;
SipDialog *gRegisterDialog = NULL;
// Only the SipMTInviteServerTransactionLayer and SipMOInviteClientTransactionLayer are allowed to call
// the underlying sipWrite method directly for the invite transactions.
void SipDialogBase::sipWrite(SipMessage *sipmsg)
{
if (!mProxy.mipValid) {
LOG(ERR) << "Attempt to write to invalid proxy ignored, address:"<<mProxy.mipName;
return;
}
gSipInterface.siWrite(&mProxy.mipSockAddr,sipmsg);
}
SipDialog *SipDialogBase::dgGetDialog()
{
return dynamic_cast<SipDialog*>(this);
}
SipState SipDialogBase::MODSendCANCEL(TermCause cause)
{
LOG(INFO) << "SIP term info MODSendCANCEL cause: " << cause;
LOG(INFO) << sbText();
setSipState(MODCanceling); // (pat) MOD sent a cancel.
SipMOCancelTU *cancelTU = new SipMOCancelTU(dynamic_cast<SipDialog*>(this),cause.tcGetSipReasonHeader());
//cancelTU->mstOutRequest.addCallTerminationReasonSM(CallTerminationCause::eQ850, cause.tcGetCCCause(), ""); // MODSendCANCEL
cancelTU->sctStart();
return getSipState();
}
void SipDialogBase::initRTP()
{
SdpInfo sdp;
sdp.sdpParse(getSdpRemote().c_str());
initRTP1(sdp.sdpHost.c_str(),sdp.sdpRtpPort,mDialogId);
}
string SipDialogBase::makeSDPOffer()
{
SdpInfo sdp;
sdp.sdpInitOffer(this);
return sdp.sdpValue();
//return makeSDP("0","0");
}
// mCodec is an implicit parameter, consisting of the chosen codec.
string SipDialogBase::makeSDPAnswer()
{
SdpInfo answer;
answer.sdpInitOffer(this);
mSdpAnswer = answer.sdpValue();
return mSdpAnswer;
}
void SipDialogBase::MTCInitRTP()
{
initRTP();
}
void SipDialogBase::MOCInitRTP()
{
initRTP();
}
void SipDialogBase::sdbText(std::ostringstream&os, bool verbose) const
{
sbText(os);
if (verbose || IS_LOG_LEVEL(DEBUG)) {
rtpText(os);
//os << "proxy=(" << mProxy.ipToText() << ")";
}
}
string SipDialogBase::sdbText() const
{
std::ostringstream ss;
sdbText(ss);
return ss.str();
}
void SipMOInviteClientTransactionLayer::MOUssdSendINVITE(string ussd, const L3LogicalChannel *chan)
{
static const char* xmlUssdTemplate =
"<?xml version=\"1.0\" encoding=\"UTF-8\"?>\n"
"<ussd-data>\n"
" <language>en</language>\n"
" <ussd-string>%s</ussd-string>\n"
"</ussd-data>\n";
LOG(INFO) << "user " << sipLocalUsername() << " state " << getSipState() <<sdbText();
static const string cInviteStr("INVITE");
SipMessage *invite = makeInitialRequest(cInviteStr);
// This is dumber than snot. We have to put in a dummy sdp with port 0.
mRTPPort = 0;
invite->smAddBody(string("application/sdp"),makeSDPOffer());
// Add RFC-4119 geolocation XML to content area, if available.
// TODO: This makes it a multipart message, needs to be tested.
string xml = format(xmlUssdTemplate,ussd);
invite->smAddBody(string("application/vnd.3gpp.ussd+xml"),xml);
SipCallbacks::writePrivateHeaders(invite,chan);
moWriteLowSide(invite);
LOG(DEBUG) <<LOGVAR(invite);
delete invite;
setSipState(Starting);
}
//old args: const char * calledUser, const char * calledDomain, short rtpPort, Control::CodecSet codec,
void SipMOInviteClientTransactionLayer::MOCSendINVITE(const L3LogicalChannel *chan)
{
static const char* xmlGeoprivTemplate =
"<?xml version=\"1.0\" encoding=\"UTF-8\"?>\n"
"<presence xmlns=\"urn:ietf:params:xml:ns:pidf\"\n"
"xmlns:gp=\"urn:ietf:params:xml:ns:pidf:geopriv10\"\n"
"xmlns:gml=\"urn:opengis:specification:gml:schema-xsd:feature:v3.0\"\n"
"entity=\"pres:%s@%s\">\n"
"<tuple id=\"1\">\n"
"<status>\n"
"<gp:geopriv>\n"
"<gp:location-info>\n"
"<gml:location>\n"
"<gml:Point gml:id=\"point1\" srsName=\"epsg:4326\">\n"
"<gml:coordinates>%s</gml:coordinates>\n"
"</gml:Point>\n"
"</gml:location>\n"
"</gp:location-info>\n"
"<gp:usage-rules>\n"
"<gp:retransmission-allowed>no</gp:retransmission-allowed>\n"
"</gp:usage-rules>\n"
"</gp:geopriv>\n"
"</status>\n"
"</tuple>\n"
"</presence>\n";
LOG(INFO) << "user " << sipLocalUsername() << " state " << getSipState() <<sdbText();
#if PAT_TEST_SIP_DIRECT
// (pat 7-23-2013): This code has eroded beyond recoverability...
//bool directBtsConnection = false;
// exten => _X.,1,Set(Name=${ODBC_SQL(select dial from dialdata_table where exten=\"${EXTEN}\")})
// exten => _X.,n,GotoIf($["${Name}"=""] ?other-lines,${EXTEN},1)
// exten => _X.,n,Set(IPAddr=${ODBC_SQL(select ipaddr from sip_buddies where username=\"${Name}\")})
// exten => _X.,n,GotoIf($["${IPAddr}"=""] ?other-lines,${EXTEN},1)
// exten => _X.,n,Set(Port=${ODBC_SQL(select port from sip_buddies where username=\"${Name}\")})
// exten => _X.,n,GotoIf($["${Port}"!=""] ?dialNum)
// exten => _X.,n,Set(Port=5062) ; Port was not set, so set to default. Gets around bug in subscriberRegistry
// exten => _X.,n(dialNum),Dial(SIP/${Name}@${IPAddr}:${Port})
if (gConfig.getStr("SIP.Proxy.Mode") == string("direct")) {
// Is this IMSI registered directly on a BTS?
string remoteIMSI = gSubscriberRegistry.getIMSI(wCalledUsername);
string remoteIPStr, remotePortStr;
if (remoteIMSI != "") {
remoteIPStr = gSubscriberRegistry.imsiGet(remoteIMSI,"ipaddr");
remotePortStr = gSubscriberRegistry.imsiGet(remoteIMSI,"port");
unsigned remotePort = (remotePortStr != "") ? atoi(remotePortStr.c_str()) : 0;
LOG(DEBUG) << "BTS direct test: "<<wCalledUsername
<< format(" -> SIP/%s@%s:%s",remoteIMSI.c_str(),remoteIPStr.c_str(),remotePortStr.c_str()) <<sdbText();
if (remoteIPStr != "" && remotePort) {
//directBTSConnection = true;
mRemoteUsername = remoteIMSI;
mProxyIP = remoteIPStr;
mProxyPort = remotePort;
LOG(INFO) << "Calling BTS direct: "<<wCalledUsername
<< format(" -> SIP/%s@%s:%u",mRemoteUsername.c_str(),mProxyIP.c_str(),mProxyPort) <<sdbText();
}
}
}
//mRemoteUsername = "IMSI001690000000002"; // pats iphone
#endif
LOG(DEBUG) <<sdbText();
static const string cInviteStr("INVITE");
SipMessage *invite = makeInitialRequest(cInviteStr);
invite->smAddBody(string("application/sdp"),makeSDPOffer());
string username = sipLocalUsername();
WATCH("MOC imsi="<<username);
if (this->dsPAssociatedUri.size()) {
invite->smAddHeader("P-Associated-URI",this->dsPAssociatedUri);
}
if (this->dsPAssertedIdentity.size()) {
invite->smAddHeader("P-Asserted-Identity",this->dsPAssertedIdentity);
}
SipCallbacks::writePrivateHeaders(invite,chan);
moWriteLowSide(invite);
delete invite;
setSipState(Starting);
};
void SipMOInviteClientTransactionLayer::handleSMSResponse(SipMessage *sipmsg)
{
// There are only three cases, and here they are:
stopTimers();
int code = sipmsg->smGetCode();
if (code < 200) {
// (pat) 11-26 This was wrong: mTimerBF.set(64*T1);
return; // Ignore 100 Trying.
}
dialogPushState((code/100)*100 == 200 ? Cleared : SSFail,code);
mTimerK.setOnce(T4); // 17.1.2.2: Timer K Controls when the dialog is destroyed.
}
void SipInviteServerTransactionLayerBase::SipMTCancel(SipMessage *sipmsg)
{
assert(sipmsg->isCANCEL());
SipMessageReply cancelok(sipmsg,200,string("OK"),this);
sipWrite(&cancelok);
if (dsPeer()->ipIsReliableTransport()) {
// It no longer matters whether we use Canceled or MTDCanceling state, and we should get rid of some states.
dialogPushState(Canceled,0);
} else {
dialogPushState(MTDCanceling,0);
setTimerJ();
}
}
void SipInviteServerTransactionLayerBase::SipMTBye(SipMessage *sipmsg)
{
assert(sipmsg->isBYE());
gReports.incr("OpenBTS.SIP.BYE.In");
SipMessageReply byeok(sipmsg,200,string("OK"),this);
sipWrite(&byeok);
if (dsPeer()->ipIsReliableTransport()) {
dialogPushState(Cleared,0);
} else {
dialogPushState(MTDClearing,0);
setTimerJ();
}
}
// Incoming message from SIPInterface.
void SipMOInviteClientTransactionLayer::MOWriteHighSide(SipMessage *sipmsg)
{
int code = sipmsg->smGetCode();
LOG(DEBUG) <<LOGVAR(code) <<LOGVAR(dgIsInvite());
if (code == 0) { // It is a SIP Request. Switch based on the method.
if (sipmsg->isCANCEL()) {
mTimerBF.stop();
// I dont think the peer is supposed to do this, but lets cancel it:
SipMTCancel(sipmsg);
} else if (sipmsg->isBYE()) {
// A BYE should not be sent by the peer until dialog established, and we are supposed to send 405 error.
// but instead we are going to quietly terminate the dialog anyway.
SipMTBye(sipmsg);
} else {
// Not expecting any others.
// Must send 405 error.
LOG(WARNING)<<"SIP Message ignored:"<<sipmsg;
SipMessageReply oops(sipmsg,405,string("Method Not Allowed"),this);
sipWrite(&oops);
}
} else { // It is a SIP reply.
saveMOResponse(sipmsg); LOG(DEBUG)<<"saveResponse"<<sipmsg;
if (dgIsInvite()) { // It is INVITE
stopTimers(); // Yes we stop the timers for every possible case. TimerBF will be restarted when handleInviteResponse sends the reply.
handleInviteResponse(code,true);
} else { // It is a MESSAGE
handleSMSResponse(sipmsg);
}
}
}
// Outgoing message.
void SipMOInviteClientTransactionLayer::moWriteLowSide(SipMessage *sipmsg)
{
if (sipmsg->isINVITE() || sipmsg->isMESSAGE()) { // It cant be anything else.
if (!dsPeer()->ipIsReliableTransport()) { mTimerAE.set(2*T1); }
mTimerBF.setOnce(64*T1); // RFC3261 17.1.2.2.2 Timer F
// This assert is not true in the weird case where we resend an SMS message.
// We should not be using this code for MESSAGE in the first place - it should be a TU.
//assert(mInvite == 0);
saveInviteOrMessage(sipmsg,true);
}
sipWrite(sipmsg);
}
// (pat) Counter-intuitively, the "ACK" is a SIP Request, not a SIP Response.
// Therefore its first line includes a Request-URI, and the request-uri is also placed in the "To" field.
// RFC2234 13.1: "The procedure for sending this ACK depends on the type of response.
// For final responses between 300 and 699, the ACK processing is done in the transaction layer and follows
// one set of rules (See Section 17). For 2xx responses, the ACK is generated by the UAC core. (section 13)"
// 17.1.1.3: "The ACK request constructed by the client transaction MUST contain
// values for the Call-ID, From, and Request-URI that are equal to the
// values of those header fields in the request passed to the transport
// by the client transaction (call this the "original request"). The To
// header field in the ACK MUST equal the To header field in the
// response being acknowledged, and therefore will usually differ from
// the To header field in the original request by the addition of the
// tag parameter. The ACK MUST contain a single Via header field, and
// this MUST be equal to the top Via header field of the original
// request. The CSeq header field in the ACK MUST contain the same
// value for the sequence number as was present in the original request,
// but the method parameter MUST be equal to "ACK".
//
void SipMOInviteClientTransactionLayer::MOCSendACK()
{
assert(! mTimerAE.isActive() && ! mTimerBF.isActive());
LOG(INFO) << sdbText();
//LOG(INFO) << "user " << mSipUsername << " state " << getSipState() <<sdbText();
static const string cAckstr("ACK");
SipMessageAckOrCancel ack(cAckstr,mInvite);
ack.msmTo = *dsRequestToHeader(); // Must get the updated to-tag.
sipWrite(&ack);
// we dont care mTimerD.set(T4);
}
void SipMOInviteClientTransactionLayer::MOSMSSendMESSAGE(const string &messageText, const string &contentType)
{
LOG(INFO) << "SIP send to " << dsRequestToHeader() <<" MESSAGE " << messageText <<sdbText();
assert(mDialogType == SIPDTMOSMS);
gReports.incr("OpenBTS.SIP.MESSAGE.Out");
static const string cMessagestr("MESSAGE");
SipMessage *msg = makeInitialRequest(cMessagestr);
msg->smAddBody(contentType,messageText);
moWriteLowSide(msg);
delete msg;
setSipState(MOSMSSubmit);
}
// Return TRUE to remove the dialog.
bool SipMOInviteClientTransactionLayer::moPeriodicService()
{
if (mTimerAE.expired()) { // Resend timer.
// The HANDOVER is inbound, but the invite is outbound like MOC.
if (getSipState() == Starting || getSipState() == HandoverInbound || getSipState() == MOSMSSubmit) {
sipWrite(getInvite());
mTimerAE.setDouble();
} else {
mTimerAE.stop();
}
} else if (mTimerBF.expired() || mTimerD.expired()) { // Dialog killer timer.
// Whoops. No response from peer.
stopTimers();
dialogPushState(SSFail,0);
return true;
} else if (mTimerK.expired()) { // Normal exit delay to absorb resends.
stopTimers();
if (! dgIsInvite()) { return true; } // It is a SIP MESSAGE.
} else if (sipIsFinished()) {
// If one of the kill timers is active, wait for it to expire, otherwise kill now.
return (mTimerBF.isActive() || mTimerD.isActive() || mTimerK.isActive()) ? false : true;
}
return false;
}
string SipMOInviteClientTransactionLayer::motlText() const
{
ostringstream os;
os <<LOGVAR2("Timers: AE",mTimerAE) <<LOGVAR2("BF",mTimerBF) <<LOGVAR2("K",mTimerK) <<LOGVAR2("D",mTimerD);
return os.str();
}
void SipMTInviteServerTransactionLayer::MTCSendTrying()
{
SipMessage *invite = getInvite();
if (invite==NULL) {
setSipState(SSFail);
gReports.incr("OpenBTS.SIP.Failed.Local");
}
LOG(INFO) << sdbText();
if (getSipState()==SSFail) return;
SipMessageReply trying(invite,100,string("Trying"),this);
mtWriteLowSide(&trying);
setSipState(Proceeding);
}
void SipMTInviteServerTransactionLayer::MTSMSSendTrying()
{
MTCSendTrying();
}
void SipMTInviteServerTransactionLayer::MTCSendRinging()
{
if (getSipState()==SSFail) return;
SipMessage *invite = getInvite();
LOG(INFO) <<sdbText();
assert(invite);
LOG(DEBUG) << "send ringing" <<sdbText();
SipMessageReply ringing(invite,180,string("Ringing"),this);
mtWriteLowSide(&ringing);
setSipState(Proceeding);
}
void SipMTInviteServerTransactionLayer::MTCSendOK(CodecSet wCodec, const L3LogicalChannel *chan)
{
if (getSipState()==SSFail) { devassert(0); }
SipMessage *invite = getInvite();
gReports.incr("OpenBTS.SIP.INVITE-OK.Out");
mRTPPort = allocateRTPPorts();
mCodec = wCodec;
LOG(INFO) <<sdbText();
SipMessageReply ok(invite,200,string("OK"),this);
ok.smAddBody(string("application/sdp"),makeSDPAnswer()); // TODO: This should be a reply to the originating SDP offer.
// (pat) Chan is NULL when in a weird special handled in dialogCancel.
if (chan) SipCallbacks::writePrivateHeaders(&ok,chan);
mtWriteLowSide(&ok);
setSipState(Connecting);
// In RFC-3261 the Transaction Layer no longer handles timers after the OK is sent.
// The Transport Layer alone is not capabable of sending the 200 OK reliably because then the
// INVITE server transaction ends, and the INVITE client transaction no longer resends INVITEs after
// receiving a provisional response. Rather, the way that would end up being handled is by starting a new
// INVITE transaction, which is totally not what we want to do. So we will push out the 2xx OK
// until we get the ACK. Doesnt matter for reliable transports.
if (dgIsInvite()) { setTimerG(); }
setTimerH();
}
string SipMTInviteServerTransactionLayer::mttlText() const
{
ostringstream ss;
ss << LOGVAR2("Timers: G",mTimerG) <<LOGVAR2("H",mTimerH) <<LOGVAR2("J",mTimerJ);
return ss.str();
}
// Doesnt seem like messages need the private headers.
void SipMTInviteServerTransactionLayer::MTSMSReply(int code, const char *explanation) // , const L3LogicalChannel *chan)
{
LOG(INFO) <<sdbText();
// If this operation was initiated from the CLI, there was no MESSAGE
if (mInvite) { // It is a MESSAGE in this case, not an INVITE
//2-2014: the reply to MESSAGE must include the to-field, so we pass the dialog to SIpMessageReply
SipMessageReply reply(mInvite,code,string(explanation),this); // previous: NULL);
sipWrite(&reply);
} else {
LOG(INFO) << "clearing CLI-generated transaction" <<sdbText();
}
setSipState(code == 200 ? Cleared : SSFail);
}
// This can only be used for early errors before we get the ACK.
void SipMTInviteServerTransactionLayer::MTCEarlyError(TermCause cause) // The message must be 300-699.
{
LOG(DEBUG) << LOGVAR(cause);
string reason;
int sipcode = cause.tcGetSipCodeAndReason(reason);
devassert(sipcode);
// Double check cause validity.
if (sipcode == 0) { // This should never happen; this is just a last resort in case of bugs.
sipcode = 486; reason = "No_User_Responding";
}
// TODO: What if we were already ACKed?
setSipState(MODError);
SipMessageReply reply(getInvite(),sipcode,reason,this); // The message must be 300-699.
reply.msmReasonHeader = cause.tcGetSipReasonHeader(); // Returns a SIP "Reason:" header string.
mtWriteLowSide(&reply);
if (dgIsInvite()) { setTimerG(); }
setTimerH();
}
// This is called for the second and subsequent received INVITEs as well as the ACK.
// We send the current response, whatever it is.
void SipMTInviteServerTransactionLayer::MTWriteHighSide(SipMessage *sipmsg) { // Incoming message from SIPInterface.
LOG(DEBUG);
SipState state = getSipState();
if (sipmsg->smGetCode() == 0) {
if (sipmsg->isINVITE()) {
if (mtLastResponse.smIsEmpty()) { MTCSendTrying(); }
else { sipWrite(&mtLastResponse); }
} else if (sipmsg->isACK()) {
if (state == SSNullState || state == Proceeding || state == Connecting) {
dialogPushState(Active,0);
gSipInterface.dmAddLocalTag(dgGetDialog());
} else {
// We could be failed or canceled, so ignore the ACK.
}
stopTimers(); // happiness
// The spec shows a short Timer I being started here, but all it does is specify a time
// when the dialog will stop absorbing additional ACKS, thus suppressing error messages.
// Well, how about if we just never throw an error for that? Done.
} else if (sipmsg->isCANCEL()) {
SipMTCancel(sipmsg);
} else if (sipmsg->isBYE()) { // TODO: This should not happen.
SipMTBye(sipmsg);
} else if (sipmsg->isMESSAGE()) {
// It is a duplicate MESSAGE. Resend the current response.
if (mtLastResponse.smIsEmpty()) {
if (! gConfig.getBool("SIP.RFC3428.NoTrying")) { MTSMSSendTrying(); } // Otherwise just ignore the duplicate MESSAGE.
} else {
sipWrite(&mtLastResponse);
}
} else {
// Not expecting any others. Must send 405 error.
LOG(WARNING)<<"SIP Message ignored:"<<sipmsg;
SipMessageReply oops(sipmsg,405,string("Method Not Allowed"),this);
sipWrite(&oops);
}
} else {
LOG(WARNING)<<"SIP Message ignored:"<<sipmsg;
}
}
// Return TRUE to remove the dialog.
bool SipMTInviteServerTransactionLayer::mtPeriodicService()
{
if (mTimerG.expired()) { // Resend timer.
if (getSipState() == SSFail || getSipState() == Active) {
sipWrite(mLastResponse);
mTimerG.setDouble(T2); // Will send again later.
} else {
// This could happen if a CANCEL started before the ACK was received.
// Not sure what to do - I think we will let the CANCEL take precedence, so stop sending this response.
mTimerG.stop();
}
} else if (mTimerJ.expired() || mTimerH.expired()) { // Dialog killer timers.
stopTimers(); // probably redundant.
// Time to destroy the Dialog.
if (dgIsInvite()) {
// Whoops. No ACK received. Notify L3 and remove the dialog
dialogPushState(SSFail,0);
} else {
// No need to notify, just remove the dialog.
}
return true; // Stop the dialog now. It will be deleted by the periodic service loop after the associated L3 transaction ends.
} else if (sipIsFinished()) {
// If one of the kill timers is active, wait for it to expire, otherwise kill now.
return (mTimerJ.isActive() || mTimerH.isActive()) ? false : true;
}
return false;
}
SipDialog *getRegistrar()
{
if (gRegisterDialog == NULL) {
gRegisterDialog = SipDialog::newSipDialogRegister1();
} else {
// This allows the user to change SIP.Proxy.Registration from the CLI.
gRegisterDialog->updateProxy("SIP.Proxy.Registration");
}
return gRegisterDialog;
}
// We wrap our REGISTER messages inside a dialog object, even though it is technically not a dialog.
SipMessage *SipDialog::makeRegisterMsg(DialogType wMethod, const L3LogicalChannel* chan, string RAND, const FullMobileId &msid, const char *SRES)
{
// TODO: We need to make a transaction here...
static const string registerStr("REGISTER");
// The request URI is special for REGISTER;
string reqUri = string("sip:") + proxyIP();
// We formerly allocated a new Dialog for each REGISTER message so the IMSI was stashed in the dialog, and localSipUri() worked.
//SipPreposition myUri(localSipUri());
string username = msid.fmidUsername();
// RFC3261 is somewhat contradictory on the From-tag and To-tag.
// The documentation for 'from' says the from-tag is always included.
// The examples in 24.1 show a From-tag but no To-tag.
// The To-tag includes the optional <>, and Paul at null team incorrectly thought the <> were required,
// so we will include them as that appears to be common practice.
string myUriString;
string authUri;
string authUsername;
string realm = gConfig.getStr("SIP.Realm");
if (realm.length() > 0) {
authUri = string("sip:") + realm;
authUsername = string("IMSI") + msid.mImsi;
myUriString = makeUri(username,realm,0);
} else {
myUriString = makeUri(username,dsPeer()->mipName,0); // The port, if any, is already in mipName.
}
//string fromUriString = makeUriWithTag(username,dsPeer()->mipName,make_tag()); // The port, if any, is already in mipName.
SipPreposition toHeader("",myUriString,"");
SipPreposition fromHeader("",myUriString,make_tag());
dsNextCSeq(); // Advance CSeqNum.
SipMessage *msg = makeRequest(registerStr,reqUri,username,&toHeader,&fromHeader,make_branch());
// Replace the Contact header so we can set the expires option. What a botched up spec.
// Replace the P-Preferred-Identity
unsigned expires;
if (wMethod == SIPDTRegister ) {
expires = 60*gConfig.getNum("SIP.RegistrationPeriod");
if (SRES && strlen(SRES)) {
if (realm.length() > 0) {
string response = makeResponse(authUsername, realm, SRES, registerStr, authUri, RAND);
msg->msmAuthorizationValue = format("Digest realm=\"%s\", username=\"%s\", nonce=\"%s\", uri=\"%s\", response=\"%s\", algorithm=MD5, qop=\"auth\" ",
realm.c_str(), authUsername.c_str(), RAND.c_str(), authUri.c_str(), response.c_str());
} else {
// (pat 9-2014) These fields are all supposed to be quoted. It was a mistake not to quote them.
// The above code that quotes the fields was used for kazoo, so the unquoted strings are only use for sipauthserve.
// Also remove the extraneous leading comma.
// RFC3261 says the fields are quoted.
//msg->msmAuthorizationValue = format("Digest, nonce=%s, uri=%s, response=%s",RAND.c_str(),msid.mImsi.c_str(),SRES);
msg->msmAuthorizationValue = format("Digest nonce=\"%s\", uri=\"%s\", response=\"%s\"",RAND.c_str(),msid.mImsi.c_str(),SRES);
}
}
} else if (wMethod == SIPDTUnregister ) {
expires = 0;
} else { assert(0); }
// We use myURI instead of localContact because the SIPDialog for REGISTER is shared by all REGISTER
// users and does not contain the personal info for this user.
//msg->msmContactValue = format("<%s>;expires=%u",myUriString.c_str(),expires);
msg->msmContactValue = localContact(username,expires);
SipCallbacks::writePrivateHeaders(msg,chan);
return msg;
}
void SipDialog::dgReset()
{
mPrevDialogState = DialogState::dialogUndefined; sipStopTimers();
//mDownlinkFifo.clear();
}
void SipDialog::MODSendBYE(TermCause cause)
{
LOG(INFO) <<sdbText();
LOG(INFO) << "SIP term info MODSendBYE cause: " << cause; // SVGDBG
setSipState(MODClearing);
SipMOByeTU *byeTU = new SipMOByeTU(this,cause.tcGetSipReasonHeader());
//byeTU->mstOutRequest.addCallTerminationReasonSM(CallTerminationCause::eQ850, cause.tcGetCCCause(), ""); // MODSendBYE
byeTU->sctStart();
}
void SipDialog::sendInfoDtmf(unsigned bcdkey)
{
// Has a previous DTMF not finished yet?
// Start a new Sip INFO Transaction to send the key off.
SipDtmfTU *dtmfTU = new SipDtmfTU(this,bcdkey);
dtmfTU->sctStart();
}
// (pat) This is the post-l3-rewrite way, most initialization during construction.
SipDialog *SipDialog::newSipDialogMT(DialogType dtype, SipMessage *req)
{
LOG(DEBUG);
assert(dtype == SIPDTMTC || dtype == SIPDTMTSMS);
string proxy = req->smGetProxy(); // Get it from the top via.
if (proxy.empty()) {
LOG(ERR) << "Missing proxy (from top via) in MT SIP message:"<<req;
// Guess at a proxy and try to keep going.
proxy = gConfig.getStr(dtype == SIPDTMTSMS ? "SIP.Proxy.SMS" : "SIP.Proxy.Speech");
}
SipDialog *dialog = new SipDialog(dtype,proxy,"INVITE or MESSAGE via");
// 2-2014: RFC 3267 8.2.6.2 says the UAS (sip server) MUST add a "to" tag to a response, and MAY add a "to" tag to a provisional (100) response.
// The reason is in case the request is forked, the client could distinguish responses from multiple servers, a case that would not happen for us.
dialog->dsSetLocalHeaderMT(&req->msmTo,dtype == SIPDTMTC);
dialog->dsSetRemoteHeader(&req->msmFrom);
//dialog->mSipUsername = req->smUriUsername(); // IMSI/TMSI is in both the URI and the To: header.
// TODO: Validate username - must be valid IMSI or TMSI.
ScopedLock lock(dialog->mDialogLock,__FILE__,__LINE__); // probably unnecessary.
dialog->dsSetCallId(req->msmCallId);
dialog->mSdpOffer = req->msmBody; // Only useful for MTC, a no-op for MTSMS.
dialog->saveInviteOrMessage(req,false);
gSipInterface.dmAddCallDialog(dialog);
return dialog;
}
// There is just one SipDialog that handles all REGISTER requests.
SipDialog *SipDialog::newSipDialogRegister1() // caller imsi
{
LOG(DEBUG);
SipDialog *dialog = new SipDialog(SIPDTRegister,gConfig.getStr("SIP.Proxy.Registration"),"SIP.Proxy.Registration");
// RFC3261 10.2: REGISTER fields are different from normal requests.
// The Request URL is the IP address (only) of the Registrar.
// The To: is the 'address of record' formatted as a SIP URI.
// The From: is the 'responsible party' and is equal to To: unless it is a third-party registration.
// What about tags? I dont think it needs them because it is not a dialog creating request, but we add them
// anyway and it hasn't hurt anything.
dialog->dsSetCallId(globallyUniqueId(""));
gSipInterface.dmAddCallDialog(dialog);
return dialog;
}
// Open an MOSMS [Mobile Originated Short Message Service] SIP Transaction and send the invite.
// We use a dialog for this even though it is just a message because it was easier to interface
// to the Control directory without changing anything.
SipDialog *SipDialog::newSipDialogMOSMS(
TranEntryId tranid,
const FullMobileId &fromMsId, // caller imsi
const string &calledDigits, // number being called, or it may be config option SIP.SMSC
const string &body,
const string &contentType)
{
LOG(DEBUG) <<LOGVAR(fromMsId)<<LOGVAR2("called",calledDigits); //<<LOGVAR2("tranid",wTranId);
// This is weird - use the local IP address as the domain of the remote user?
SipDialog *dialog = new SipDialog(SIPDTMOSMS,gConfig.getStr("SIP.Proxy.SMS"),"SIP.Proxy.SMS");
dialog->dsSetLocalMO(fromMsId, true);
string calledDomain = dialog->localIP();
dialog->dsSetRemoteUri(makeUri(calledDigits,calledDomain));
dialog->smsBody = body; // Temporary until smqueue is fixed.
dialog->smsContentType = contentType; // Temporary until smqueue is fixed.
// Must lock once we do dmAddCallDialog to prevent the SIPInterface threads from accessing this dialog
// while we finish construction.
ScopedLock lock(dialog->mDialogLock,__FILE__,__LINE__);
gSipInterface.dmAddCallDialog(dialog);
//dialog->MOSMSSendMESSAGE(calledDigits,calledDomain,body,contentType);
gNewTransactionTable.ttSetDialog(tranid,dialog); // Must do this before the dialog receives any messages.
dialog->MOSMSSendMESSAGE(body,contentType);
return dialog;
}
// SIP-URI = "sip:" [ userinfo ] hostport
// userinfo = ( user / telephone-subscriber ) [ ":" password ] "@"
// user = 1*( unreserved / escaped / user-unreserved )
// unreserved = alphanum / mark
// mark = "-" / "_" / "." / "!" / "~" / "*" / "'" / "(" / ")"
// escaped = "%" HEXDIG HEXDIG
// user-unreserved = "&" / "=" / "+" / "$" / "," / ";" / "?" / "/"
// Any other character needs to be escaped. RFC 2396.
//
// In general, URI encoding (RFC2396) is kind of complicated - for example, we need a-priori knowledge if ";" in the string is
// marking a URI parameter, in which case it needs to be left alone, or if it is part of the URI component, in which case it must be escaped.
// RFC3267 talks about URBNF the URI string may contain:
// This handles the USSD special case. A USSD request can contain only digits, letters * and #,
// and of these the only special one is "#" turns into "%23".
string escapeUssdUri(string ss)
{
size_t pos = 0;
while (1) {
pos = ss.find(pos,'#');
if (pos == string::npos) return ss;
ss.replace(pos,1,"%23"); // it is ok not to advance pos because replacement "%23" does not include search "#".
}
return ss;
}
SipDialog *SipDialog::newSipDialogMOUssd(
TranEntryId tranid,
const FullMobileId &fromMsId, // caller imsi
const string &wUssd, // USSD string entered by user to send to network.
L3LogicalChannel *chan
)
{
LOG(DEBUG) << "MOUssd (INVITE)"<<LOGVAR(fromMsId)<<LOGVARM(wUssd);
// TODO: The SIPEngine constructor calls sipSetUser. FIX IT. Maybe I just need to replace SIPEngine.
const char *proxyOption = "SIP.Proxy.USSD";
string proxy = gConfig.getStr(proxyOption);
LOG(DEBUG) << LOGVAR(proxyOption) <<LOGVAR(proxy);
if (proxy.length() > 259) { // TODO: This should be in the config checker, if anywhere.
LOG(ALERT) << "Configured " <<proxyOption <<" hostname is greater than 253 bytes!";
}
SipDialog *dialog = new SipDialog(SIPDTMOUssd,proxy,proxyOption);
dialog->dsSetLocalMO(fromMsId,true);
gReports.incr("OpenBTS.SIP.INVITE.Out");
// Must lock once we do dmAddCallDialog to prevent the SIPInterface threads from accessing this dialog
// while we finish construction.
ScopedLock lock(dialog->mDialogLock,__FILE__,__LINE__); // Must lock before dmAddCallDialog.
if (proxy == "testmode") {
gNewTransactionTable.ttSetDialog(tranid,dialog); // Must do this before the dialog receives any messages.
DialogUssdMessage *dmsg = new DialogUssdMessage(tranid,DialogState::dialogBye,0);
dmsg->dmMsgPayload = "Hello from OpenBTS. You entered:"+wUssd;
LOG(DEBUG) << "USSD test mode"<<LOGVAR(chan)<<LOGVAR(tranid)<<LOGVAR(fromMsId)<<dmsg->dmMsgPayload;
dialog->dialogQueueMessage(dmsg);
return dialog;
}
// Yes, you really do put the USSD request in the To: of the SIP as per 3GPP 24.090 Appendix A, but we need to
// escape the "#" character. We also add a "dialstring" tag.
// Like this: INVITE sip:*135%23;phone-context=home1.net;user=dialstring SIP/2.0
// The obvious way to pass a USSD string is as a "tel:" domain, but evidently that was too easy, so we
// must add the user=dialstring tag as per RFC 4967, and then THAT requires a "context" as per RFC3966 because
// this is not a "global" tel number, and therefore by definition it must be a "local" tel number, which requires
// a context, but we learn from RFC 4967 that we can stick in anything we want for the context, since it is meaningless
// when used for a USSD string. This spec is not exactly an "A-team" effort.
string ussdEscaped = escapeUssdUri(wUssd);
string ussdPerSpec = ussdEscaped + ";phone-context=irrelevant.net;user=dialstring";
dialog->dsSetRemoteUri(makeUri(ussdPerSpec,dialog->localIP()));
// TODO: What about codecs? The example in 24.390 annex A has them.
gSipInterface.dmAddCallDialog(dialog);
gNewTransactionTable.ttSetDialog(tranid,dialog); // Must do this before the dialog receives any messages.
dialog->MOUssdSendINVITE(wUssd,chan);
return dialog;
}
// Open an MOC [Mobile Originated Call] dialog and send the invite.
SipDialog *SipDialog::newSipDialogMOC(
TranEntryId tranid,
const FullMobileId &fromMsId, // caller imsi
const string &wCalledDigits, // number being called, or empty for an emergency call.
CodecSet wCodecs, // phone capabilities
L3LogicalChannel *chan
)
{
LOG(DEBUG) << "MOC SIP (INVITE)"<<LOGVAR(fromMsId)<<LOGVAR2("called",wCalledDigits);
// TODO: The SIPEngine constructor calls sipSetUser. FIX IT. Maybe I just need to replace SIPEngine.
const char *proxyOption = "SIP.Proxy.Speech";
string proxy = gConfig.getStr(proxyOption);
LOG(DEBUG) << LOGVAR(proxyOption) <<LOGVAR(proxy);
if (proxy.length() > 259) { // TODO: This should be in the config checker, if anywhere.
LOG(ALERT) << "Configured " <<proxyOption <<" hostname is greater than 253 bytes!";
}
SipDialog *dialog = new SipDialog(SIPDTMOC,proxy,proxyOption);
dialog->dsSetLocalMO(fromMsId,true);
{
gReports.incr("OpenBTS.SIP.INVITE.Out");
dialog->dsSetRemoteUri(makeUri(wCalledDigits,dialog->localIP()));
}
string username = fromMsId.fmidUsername();
if (username.length()) {
devassert(username.substr(0,4) == "IMSI");
string pAssociatedUri, pAssertedIdentity;
gTMSITable.getSipIdentities(username.substr(4),pAssociatedUri,pAssertedIdentity); // They may be empty.
dialog->dsPAssociatedUri = pAssociatedUri;
dialog->dsPAssertedIdentity = pAssertedIdentity;
WATCH("MOC"<<LOGVAR(username)<<LOGVAR(pAssociatedUri)<<LOGVAR(pAssertedIdentity));
//if (pAssociatedUri.size()) { invite->smAddHeader("P-Associated-URI",pAssociatedUri); }
//if (pAssertedIdentity.size()) { invite->smAddHeader("P-Asserted-Identity",pAssertedIdentity); }
}
dialog->mRTPPort = Control::allocateRTPPorts();
dialog->mCodec = wCodecs;
// Must lock once we do dmAddCallDialog to prevent the SIPInterface threads from accessing this dialog
// while we finish construction.
ScopedLock lock(dialog->mDialogLock,__FILE__,__LINE__); // Must lock before dmAddCallDialog.
gSipInterface.dmAddCallDialog(dialog);
gNewTransactionTable.ttSetDialog(tranid,dialog); // Must do this before the dialog receives any messages.
dialog->MOCSendINVITE(chan);
return dialog;
}
// This is called in BS2 after a handover complete is received. It is an inbound handover, but an outoing MO re-INVITE.
// We take the SIP REFER message created by BS1 and send it to the SIP server as a re-INVITE.
// Note that the MS may go from BS1 to BS2 and back to BS1, in which case there may
// already be an existing dialog in some non-Active state.
SipDialog *SipDialog::newSipDialogHandover(TranEntry *tran, string sipReferStr)
{
LOG(DEBUG)<<LOGVAR(tran) <<LOGVAR(sipReferStr);
static const string inviteStr("INVITE");
// Init the Dialog State from the SIP REFER message.
SipMessage *msg = sipParseBuffer(sipReferStr.c_str());
if (msg == NULL) { return NULL; } // Message already printed.
SipUri referto(msg->msmHeaders.paramFind("Refer-To"));
string proxy = referto.uriHostAndPort();
// 7-23 wrong: SipDialog *dialog = new SipDialog(SIPDTMTC,proxy);
SipDialog *dialog = new SipDialog(SIPDTMOC,proxy,"REFER message");
dialog->mIsHandover = true;
dialog->dsSetRemoteHeader(&msg->msmTo);
dialog->dsSetLocalHeader(&msg->msmFrom);
dialog->dsSetCallId(msg->msmCallId);
// TODO: If any other intervening messages were sent by BTS1 between the REFER and now the CSeqNum will not be correct.
dialog->mLocalCSeq = msg->msmCSeqNum + 1;
// We copied the peer SDP we got from the SIP server into the handover message passed from BS1 to BS2;
// I dont think we need to save sdpResponse here - we are going to use it for the last time immediately below.
dialog->mCodec = tran->getCodecs(); // TODO: We need to renegotiate this, or set it from SDP. There is no point even setting this here.
// Get remote RTP from SIP REFER message, init RTP, create new SDP offer from previous SDP response.
// The incoming SDP has the codec previously negotiated, so it should still be ok.
dialog->mRTPPort = Control::allocateRTPPorts();
SdpInfo sdpRemote;
sdpRemote.sdpParse(msg->msmBody);
SdpInfo sdpLocal = sdpRemote; // In particular, we are copying the sessionId and versionId.
// Send our local RTP port to the SIP server.
sdpLocal.sdpRtpPort = dialog->mRTPPort;
sdpLocal.sdpHost = dialog->localIP();
dialog->mSdpOffer = sdpLocal.sdpValue();
// Make the re-INVITE
SipMessage *invite = dialog->makeInitialRequest(inviteStr);
invite->smAddBody(string("application/sdp"),dialog->mSdpOffer);
// Send it off.
ScopedLock lock(dialog->mDialogLock,__FILE__,__LINE__);
gSipInterface.dmAddCallDialog(dialog);
dialog->moWriteLowSide(invite);
delete invite; // moWriteLowSide saved a copy of this.
dialog->setSipState(HandoverInbound);
return dialog;
}
SipDialog::~SipDialog()
{
// nothing
}
TranEntry *SipDialog::findTranEntry()
{
if (this->mTranId == 0) {
// No attached transaction. Can happen if we jumped the gun (the dialog is created before the transaction
// and there could be a race with an incoming message) or if we responded with an early error
// to a dialog and never created a transaction for it, for example, 486 Busy Here.
return NULL;
}
return gNewTransactionTable.ttFindById(this->mTranId);
}
// If it is not an IMSI we think it may be a phone number.
static bool isPhoneNumber(string thing)
{
if (thing.size() == 0) { return false; } // Not a phone number.
if (0 == strncasecmp(thing.c_str(),"IMSI",4)) { return false; } // It is an IMSI, not a phone number.
return true; // Well, maybe it is a phone number.
}
static string removeUriFluff(string thing)
{
if (unsigned first = thing.find('<') != string::npos) { // Remove the angle brackets.
thing = thing.substr(first); // chop off initial angle bracket.
thing = thing.substr(0,thing.find_last_of('>')); // chop off trailing angle bracket, and anything following.
}
thing = thing.substr(0,thing.find_last_of('@')); // Chop off the ip address, if any.
const char *str = thing.c_str();
// Very clever, that a phone number may be prefixed with either sip: or tel:
if (0 == strncasecmp(str,"sip:",4) || 0 == strncasecmp(str,"tel:",4)) {
thing = thing.substr(4);
} else if (0 == strncasecmp(str,"sips:",5)) { // secure not supported, but always hopeful....
thing = thing.substr(5);
}
LOG(DEBUG) <<LOGVAR(thing);
return thing; // Hopefully, what is remaining is a phone number.
}
TranEntry *SipDialog::createMTTransaction(SipMessage *invite)
{
TranEntry *tran = NULL;
string callerId;
string callerIdSource = gConfig.getStr("GSM.CallerID.Source");
if (0 == strcasecmp(callerIdSource.c_str(),"auto")) {
// (pat 6-2014) Added automatic caller id identification.
// We can do this automatically because we know whether the displayname, etc, are imsi because they are preceded by "IMSI".
// btw, the P-Asserted-Identity should by an IMSI; the phone number is supposed to be in the P-Associated-URI.
callerId = sipRemoteDisplayname(); // This is out best guess.
LOG(DEBUG) << "CallerID=auto: sipRemoteDisplayname="<<callerId;
if (! isPhoneNumber(callerId)) {
// Well that wasnt it, try again...
callerId = removeUriFluff(invite->msmHeaders.paramFind("P-Associated-URI"));
LOG(DEBUG) << "CallerID=auto: P-Associated-URI="<<callerId;
}
if (! isPhoneNumber(callerId)) {
// Keep trying...
callerId = removeUriFluff(invite->msmHeaders.paramFind("P-Asserted-Identity"));
LOG(DEBUG) << "CallerID=auto: P-Asserted-Identity="<<callerId;
}
if (! isPhoneNumber(callerId)) {
// The SIP username is not likely a phone number, but it is our last hope.
callerId = removeUriFluff(sipRemoteUsername());
LOG(DEBUG) << "CallerID=auto: sipRemoteUserName="<<callerId;
}
if (! isPhoneNumber(callerId)) {
callerId = string(""); // Did not find a phone number, sigh, zero it out.
LOG(DEBUG) << "CallerID=auto: giving up";
}
} else if (callerIdSource.compare("username") == 0) {
callerId = sipRemoteUsername();
LOG(INFO) << "source=username, callerId = " << callerId;
} else if (callerIdSource.compare("p-asserted-identity") == 0) {
string tmpcid = invite->msmHeaders.paramFind("P-Asserted-Identity");
unsigned first = tmpcid.find("<sip:");
unsigned last = tmpcid.find_last_of("@");
callerId = tmpcid.substr(first+5, last-first-5);
LOG(INFO) << "source=p-asserted-identity, callerId = " << callerId;
} else {
callerId = sipRemoteDisplayname();
LOG(INFO) << "source=username, callerId = " << callerId;
}
FullMobileId msid;
msid.mImsi = invite->smGetInviteImsi();
if (invite->isINVITE()) {
tran = TranEntry::newMTC(this,msid,GSM::L3CMServiceType::MobileTerminatedCall,callerId);
// Tell the sender we are trying.
this->MTCSendTrying();
} else {
devassert(0);
}
return tran;
}
// If the cause is handoverOutbound, kill the dialog now: dont send a BYE, dont wait for any other incoming messsages.
// Used for outbound handover, where the SIP session was transferred to another BTS.
// cause == This is the reason a Transaction (TranEntry) was cancelled
// Layer3 may call this multiple times just to be safe and make sure the dialog is truly cancelled;
// generally the first call includes the real causes, then the second call is made when the transaction
// is destroyed and the cause is some bogus cause.
void SipDialog::dialogCancel(TermCause cause)
{
SIP::SipState state = this->getSipState();
//bool bTerminationAdded = false;
LOG(INFO) << "SIP term info dialogCancel begin cancel cause: " << cause; // SVGDBG
WATCH("dialogCancel"<<LOGVAR(state)<<LOGVAR(cause) )
ScopedLock lock(mDialogLock,__FILE__,__LINE__);
LOG(DEBUG) << dialogText(); // "SIP state " << state; //SVGDBG switch to LOG(INFO)
if (cause.tcGetValue() == L3Cause::Handover_Outbound) {
// Terminate the dialog instantly. Dont send anything on the SIP interface.
sipStopTimers();
// We need to remove the callid of the terminated outbound dialog queue from SIPInterface in case
// the same call is handerovered back, it would then be a duplicate.
gSipInterface.dmRemoveDialog(this);
LOG(INFO) << "SIP term info dialogCancel dmRemoveDialog return";
return;
}
// why aren't we checking for failed here? -kurtis ; we are now. -david
if (this->sipIsFinished()) {
// No bye or cancel message will be sent.
LOG(INFO) << "SIP term info dialogCancel sipIsFinished return SIP state: " << this->getSipState();
return;
}
switch (mDialogType) {
case SIPDTRegister:
case SIPDTUnregister:
// The Register is not a full dialog so we dont send anything when we cancel.
break;
case SIPDTMOSMS:
case SIPDTMTSMS:
case SIPDTMOUssd:
setSipState(Cleared);
break;
case SIPDTMTC:
case SIPDTMOC:
switch (state) {
case SSTimeout:
case MOSMSSubmit: // Should never see a message state in an INVITE dialog.
LOG(ERR) "Unexpected SIP State:"<<state;
break;
case Active: // (pat) MOC received OK; MTC sent ACK
caseActive:
//Changes state to clearing
this->MODSendBYE(cause);
//bTerminationAdded = true;
//then cleared
sipStartTimers(); // formerly: MODWaitForBYEOK();
break;
case SSNullState: // (pat) MTC initial state - nothing sent yet. MOC not used because sends INVITE on construction.
case Starting: // (pat) MOC or MOSMS or inboundHandover sent INVITE; MTC not used.
case Proceeding: // (pat) MOC received Trying, Queued, BeingForwarded; MTC sent Trying
case Ringing: // (pat) MOC received Ringing, notably not used for MTC sent Ringing.
case MOCBusy: // (pat) MOC received Busy; MTC not used.
case Connecting: // (pat) MTC sent OK.
case HandoverInbound:
if (mDialogType == SIPDTMOC) {
// To cancel the invite before the ACK is received we must send CANCEL instead of BYE.
this->MODSendCANCEL(cause); //Changes state to MODCanceling
//bTerminationAdded = true;
} else {
// We are the INVITE recipient server and have not received the ACK yet, so we must send an error response.
// Way back in version 3 this was used for MTC also.
// RFC3261 (SIP) is internally inconsistent describing the error codes - the 4xxx and 5xx generic
// descriptions are contradicted by specific error code descriptions.
// This is from Paul Chitescu at Null Team:
// "A 504 Server Timeout seems the most adequate response [to MS not responding to page.]
// 408 is reserved for SIP protocol timeouts (no answer to SIP message)
// 504 indicates some other timeout beyond SIP (interworking)
// 480 indicates some temporary form of resource unavailability or congestion but
// resource is accessible and can be checked"
// 486 "Busy Here" implies that we found the MS but it really is busy.
// 503 indicates the service is unavailable but does not imply for how long
// TODO: We should probably send different codes for different reasons.
// Note: We previously sent 480.
//this->MTCEarlyError(480, "Temporarily Unavailable"); // The message must be 300-699.
if (cause.tcGetCCCause() == L3Cause::Normal_Call_Clearing) {
// (pat 7-2014) The handset MTC hung up normally but the SIP Dialog here is not in the Active state.
// This weird case occurs if the handset sends a disconnect before it sends a connect,
// which can happen on some if the user answers and hangs up immediately.
// We need to send a 200 OK and then BYE instead of sending an early error.
// We dont care about codecs because we are clearing immediately so just regurgitate the codecs from the INVITE.
LOG(DEBUG) << "SIP SPECIAL CASE: Disconnect before connect";
this->MTCSendOK(this->vGetCodecs(), NULL);
goto caseActive;
} else {
this->MTCEarlyError(cause);
}
#if PREVIOUS_CODE
// (pat) Keeping this old code here for a while to show what SIP responses were returned by the version 4 code.
// Now SIP responses are formulated from the TermCause by tcGetSipCodeAndReason()
int sipcode = 486; const char *reason = "No answer";
switch (cause) {
case CancelCauseHandoverOutbound:
case CancelCauseSipInternalError:
assert(0); // handled above
case CancelCauseNormalDisconnect: // 0 Loss of contact with MS or an error.
case CancelCauseBusy: // MS is here and unavailable.
sipcode = 486; reason = "Busy Here";
break;
case CancelCauseUnknown: // 0 Loss of contact with MS or an error.
if (l3Cause == L3Cause::SwitchingEquipmentCongestion) {
sipcode = 503; reason = "Normal circuit congestion";
}
else if (l3Cause == L3Cause::NoUserResponding) {
sipcode = 408; reason = "No user responding";
}
else if (l3Cause == L3Cause::CallRejected) {
sipcode = 603; reason = "Call rejected";
}
break;
case CancelCauseCongestion: // This reason is never used MS is here but no channel avail or other congestion.
sipcode = 503; reason = "Normal circuit congestion";
break;
case CancelCauseNoAnswerToPage: // Not used We dont have any clue if the MS is in this area or not.
// The MS is not here or turned off.
sipcode = 408; reason = "No user responding";
break;
case CancelCauseOperatorIntervention:
sipcode = 487; reason = "Request Terminated Operator Intervention";
break;
}
this->MTCEarlyError(sipcode,reason); // The message must be 300-699.
#endif
//bTerminationAdded = true;
}
break;
case MODClearing: // (pat) MOD sent BYE
case MODCanceling: // (pat) MOD sent a cancel.
case MODError: // (pat) MOD sent an error response.
case MTDClearing: // (pat) MTD received BYE.
case MTDCanceling: // (pat) MTD received CANCEL
case Canceled: // (pat) received OK to CANCEL.
case Cleared: // (pat) MTD sent OK to BYE, or MTD internal error loss of FIFO, or MOSMS received OK, or MTSMS sent OK.
case SSFail:
// Some kind of clearing already in progress. Do not repeat.
break;
case HandoverOutbound: // We never used this state.
// Not sure what to do with these.
break;
}
break;
default:
assert(0);
break;
}
LOG(INFO) << "SIP term info end dialogCancel, cancel cause: " << cause << " mDialogType: " << mDialogType <<LOGVAR(state)<<LOGVAR(cause);
// << " bTerminationAdded: " << bTerminationAdded;
// (pat) The TermCause has the complete termination reason. The cause was sent in the error response, CANCEL message, or BYE message;
// we dont need to save it in the dialog.
// Save termination reason
//if (!bTerminationAdded) {
// if (! cause.tcIsEmpty()) { // Don't log unknown reason codes
// LOG(INFO) << "SIP term info add message from dialogCancel"<<LOGVAR(cause); // SVGDBG
// addCallTerminationReasonDlg(CallTerminationCause::eQ850, cause.tcGetCCCause(), ""); // dialCancel
// }
//}
} // dialogCancel
void SipEngine::dialogQueueMessage(DialogMessage *dmsg)
{
// This was used when there was just one layer3 thread:
// TODO: We may still use this for UMTS.
//Control::gCSL3StateMachine.csl3Write(new Control::GenericL3Msg(dmsg,callID()));
// Now we enqueue dialog messages in a queue in their dialog, and let L3 fish it out from there.
// We dont enqueue on the GSM LogicalChannel because that may change from, eg, SDCCH to FACCH before this message is processed.
LOG(DEBUG) << "sending DialogMessage to L3 " /*<<dialogText()*/ <<LOGVAR(dmsg);
//mDownlinkFifo.write(dmsg);
if (mTranId == 0) {
// pat 5-2014: There will not be a transaction if the dialog was cancelled in handleInvite because the
// called user is already busy with no transaction slot available. This case will go away when
// we support call-hold/call-wait.
LOG(DEBUG) << "Warning: dialog with no attached transaction; DialogMessage ignored.";
delete dmsg;
return;
}
gNewTransactionTable.ttAddMessage(mTranId,dmsg);
}
bool SipDialog::permittedTransition(DialogState::msgState oldState, DialogState::msgState newState)
{
if (newState > oldState) { return true; } // That was easy!
if (newState == oldState) {
// Allow multiple proceeding/ringing notifications:
if (newState == DialogState::dialogProceeding || newState == DialogState::dialogRinging) { return true; }
}
return false;
}
void SipDialog::dialogPushState(
SipState newSipState, // The new sip state.
int code, // The SIP message code that caused the state change, or 0 for ACK or total failures.
char timer)
{
SipState oldSipState = getSipState();
DialogState::msgState oldDialogState = getDialogState();
setSipState(newSipState);
// If it is a new state, inform L3.
DialogState::msgState nextDialogState = getDialogState(); // based on the newSipState we just set.
LOG(DEBUG) <<LOGVAR(oldSipState)<<LOGVAR(newSipState)<<LOGVAR(getSipState())<<LOGVAR(mPrevDialogState)<<LOGVAR(oldDialogState)<<LOGVAR(nextDialogState)<<dialogText();
if (nextDialogState == DialogState::dialogStarted) {
// This state is used for MO transactions just to indicate the dialog is active,
// but the MO state machine already knows that since it created the dialog,
// so we dont return this state as a notification.
return;
}
if (permittedTransition(mPrevDialogState,nextDialogState)) {
DialogMessage *dmsg = new DialogMessage(mTranId,nextDialogState,code);
dialogQueueMessage(dmsg);
} else {
LOG(DEBUG) << "no dialog state change";
}
mPrevDialogState = nextDialogState;
// A timer may be specified if the SIP state is one indicating failure.
// The timer letter corresponds to one of those specified in RFC3261, and specifies the dialog
// should not be destroyed until the timer expires.
switch (timer) {
case 0:
break; // default, no timer specified.
case 'D':
// RFC3261 17.1.1.2 says set Timer D to 32s instead of 64*T1. Whatever.
if (dsPeer()->ipIsReliableTransport()) {
mTimerD.setOnce(32000);
}
break;
case 'K':
mTimerK.setOnce(T4);
break;
default: assert(0);
}
}
void SipDialog::dialogChangeState(
SipMessage *sipmsg) // The message that caused the state change, or NULL for total failures.
{
dialogPushState(getSipState(),sipmsg?sipmsg->smGetCode():0);
//LOG(DEBUG) <<dialogText();
//// If it is a new state, inform L3.
//DialogState::msgState nextDialogState = getDialogState();
//if (nextDialogState == DialogState::dialogStarted) {
// // This state is used for MO transactions just to indicate the dialog is active,
// // but the MO state machine already knows that since it created the dialog,
// // so we dont return this state as a notification.
// return;
//}
//if (permittedTransition(mPrevDialogState,nextDialogState)) {
// unsigned code = sipmsg ? sipmsg->smGetCode() : 0;
// DialogMessage *dmsg = new DialogMessage(mTranId,nextDialogState,code);
// // done by the register TU
// dialogQueueMessage(dmsg);
//} else {
// LOG(DEBUG) << "no dialog state change";
//}
//mPrevDialogState = nextDialogState;
}
// Only a small subset of SIP states are passed to the L3 Control layer as dialog states.
DialogState::msgState SipDialog::getDialogState() const
{
// Do not add a default case so that if someone adds a new SipState they will get a warning here.
// Therefore we define every state including the impossible ones.
switch (getSipState()) {
case SSNullState:
return DialogState::dialogUndefined;
case Starting: // (pat) MOC or MOSMS or inboundHandover sent INVITE; MTC not used.
return DialogState::dialogStarted;
case Proceeding: // (pat) MOC received Trying, Queued, BeingForwarded; MTC sent Trying
case Connecting: // (pat) MTC sent OK.
// TODO: Is this correct for MTC Connecting?
return DialogState::dialogProceeding;
case Ringing: // (pat) MOC received Ringing, notably not used for MTC sent Ringing, which is probably a bug of no import.
return DialogState::dialogRinging;
case Active: // (pat) MOC received OK; MTC sent ACK
return DialogState::dialogActive;
case MODClearing: // (pat) MOD sent BYE
case MODCanceling: // (pat) MOD sent a cancel.
case MTDClearing: // (pat) MTD received BYE.
case MTDCanceling: // (pat) received CANCEL
case Canceled: // (pat) received OK to CANCEL.
case Cleared: // (pat) MTD sent OK to BYE, or MTD internal error loss of FIFO, or MOSMS received OK, or MTSMS sent OK.
return DialogState::dialogBye;
case MOCBusy: // (pat) MOC received Busy; MTC not used.
case SSTimeout:
case MODError: // (pat) MOD sent a cancel.
case SSFail:
return DialogState::dialogFail;
//case SipRegister: // (pat) This SIPEngine is being used for registration, none of the other stuff applies.
//case SipUnregister: // (pat) This SIPEngine is being used for registration, none of the other stuff applies.
case MOSMSSubmit: // (pat) SMS message submitted, "MESSAGE" method. Set but never used.
case HandoverInbound:
case HandoverOutbound:
return DialogState::dialogUndefined;
}
devassert(0);
return DialogState::dialogUndefined;
}
// Handle response to INVITE or MESSAGE.
// Only responses (>=200) to INVITE get an ACK. Specifically, not MESSAGE.
void SipDialog::handleInviteResponse(int status,
bool sendAck) // TRUE if transaction is INVITE. We used to use this for MESSAGE also, in which case it was false.
{
LOG(DEBUG) <<LOGVAR(status) <<LOGVAR(sendAck);
switch (status) {
// class 1XX: Provisional messages
case 100: // Trying
case 181: // Call Is Being Forwarded
case 182: // Queued
case 183: // Session Progress FIXME we need to setup the sound channel (early media)
dialogPushState(Proceeding,status);
break;
case 180: // Ringing
mReceived180 = true;
dialogPushState(Ringing,status);
break;
// class 2XX: Success
case 200: // OK
// Save the response and update the state,
// but the ACK doesn't happen until the call connects.
dialogPushState(Active,status);
break;
// class 3xx: Redirection
case 300: // Multiple Choices
case 301: // Moved Permanently
case 302: // Moved Temporarily
case 305: // Use Proxy
case 380: // Alternative Service
LOG(NOTICE) << "redirection not supported code " << status <<sdbText();
dialogPushState(SSFail,status, 'D');
gReports.incr("OpenBTS.SIP.Failed.Remote.3xx");
// TODO: What if it is not MOC?
if (sendAck) MOCSendACK();
break;
// Anything 400 or above terminates the call, so we ACK.
// FIXME -- It would be nice to save more information about the
// specific failure cause.
// class 4XX: Request failures
case 405: // Method Not Allowed
// We must not ACK to "405 Method Not Allowed" or you could have an infinite loop. Saw this with smqueue.
dialogPushState(SSFail,status, 'D');
gReports.incr("OpenBTS.SIP.Failed.Remote.4xx");
break;
case 400: // Bad Request
case 401: // Unauthorized: Used only by registrars. Proxys should use proxy authorization 407
case 402: // Payment Required (Reserved for future use)
case 403: // Forbidden
case 404: // Not Found: User not found
case 406: // Not Acceptable
case 407: // Proxy Authentication Required
case 408: // Request Timeout: Couldn't find the user in time
case 409: // Conflict
case 410: // Gone: The user existed once, but is not available here any more.
case 413: // Request Entity Too Large
case 414: // Request-URI Too Long
case 415: // Unsupported Media Type
case 416: // Unsupported URI Scheme
case 420: // Bad Extension: Bad SIP Protocol Extension used, not understood by the server
case 421: // Extension Required
case 422: // Session Interval Too Small
case 423: // Interval Too Brief
case 480: // Temporarily Unavailable
case 481: // Call/Transaction Does Not Exist
case 482: // Loop Detected
case 483: // Too Many Hops
case 484: // Address Incomplete
case 485: // Ambiguous
LOG(NOTICE) << "request failure code " << status <<sdbText();
dialogPushState(SSFail,status, 'D');
gReports.incr("OpenBTS.SIP.Failed.Remote.4xx");
if (sendAck) MOCSendACK();
break;
case 486: // Busy Here
LOG(NOTICE) << "remote end busy code " << status <<sdbText();
dialogPushState(MOCBusy,status,'D');
// TODO: What if it is not MOC?
if (sendAck) MOCSendACK();
break;
case 487: // Request Terminated
case 488: // Not Acceptable Here
case 491: // Request Pending
case 493: // Undecipherable: Could not decrypt S/MIME body part
LOG(NOTICE) << "request failure code " << status <<sdbText();
dialogPushState(SSFail,status,'D');
gReports.incr("OpenBTS.SIP.Failed.Remote.4xx");
if (sendAck) MOCSendACK();
break;
// class 5XX: Server failures
case 500: // Server Internal Error
case 501: // Not Implemented: The SIP request method is not implemented here
case 502: // Bad Gateway
case 503: // Service Unavailable
case 504: // Server Time-out
case 505: // Version Not Supported: The server does not support this version of the SIP protocol
case 513: // Message Too Large
LOG(NOTICE) << "server failure code " << status <<sdbText();
dialogPushState(SSFail,status,'D');
gReports.incr("OpenBTS.SIP.Failed.Remote.5xx");
// TODO: What if it is not MOC?
if (sendAck) MOCSendACK();
break;
// class 6XX: Global failures
case 600: // Busy Everywhere
case 603: // Decline
dialogPushState(MOCBusy,status,'D');
if (sendAck) MOCSendACK();
break;
case 604: // Does Not Exist Anywhere
case 606: // Not Acceptable
LOG(NOTICE) << "global failure code " << status <<sdbText();
dialogPushState(SSFail,status,'D');
gReports.incr("OpenBTS.SIP.Failed.Remote.6xx");
if (sendAck) MOCSendACK();
default:
LOG(NOTICE) << "unhandled status code " << status <<sdbText();
dialogPushState(SSFail,status,'D');
gReports.incr("OpenBTS.SIP.Failed.Remote.xxx");
if (sendAck) MOCSendACK();
}
}
// Look for <tag>blah</tag> in xmlin and return "blah".
static string xmlFind(const char *xmlin, const char *tag)
{
char tagbuf[56];
assert(strlen(tag) < 50);
sprintf(tagbuf,"<%s>",tag);
const char *start = strstr(xmlin,tagbuf);
if (!start) return string("");
const char *result = start + strlen(tagbuf);
sprintf(tagbuf,"</%s>",tag);
const char *end = strstr(start,tagbuf);
if (!start) return string("");
return string(result,end-result);
}
// The incoming USSD BYE message could have a payload to be sent to the MS.
void SipDialog::handleUssdBye(SipMessage *msg)
{
// There could be multiple BYE messages, hopefully all identical, but we only want to send one DialogMessage.
if (getSipState() == Cleared) return;
DialogUssdMessage *dmsg = new DialogUssdMessage(mTranId,DialogState::dialogBye,0);
// Is it is ok for there to be no response string?
// We have to send something to the MS so in that case return an empty string.
if (msg->smGetMessageContentType().find("application/vnd.3gpp.ussd+xml") == string::npos) {
LOG(INFO) << "UUSD response does not contain correct body type";
} else {
dmsg->dmMsgPayload = xmlFind(msg->smGetMessageBody().c_str(),"ussd-string");
if (dmsg->dmMsgPayload == "") {
// This is ok.
LOG(INFO) << "Missing UUSD response does not contain correct body type";
}
}
dialogQueueMessage(dmsg);
if (dsPeer()->ipIsReliableTransport()) {
dialogPushState(Cleared,0);
} else {
dialogPushState(MTDClearing,0);
setTimerJ();
}
}
// The SIPInterface sends this to us based on mCallId.
// We will process the message and possibly send replies or DialogMessages to the L3 State machine.
// Blah, this should be handled by Dialog sub-classes.
void SipDialog::dialogWriteDownlink(SipMessage *msg)
{
LOG(DEBUG) <<"received SIP" /*<<LOGVAR2("SIP.state",sipState())*/ <<" msg:"<<msg->text() <<dialogText();
sipStopTimers();
ScopedLock lock(mDialogLock,__FILE__,__LINE__);
unsigned code = msg->smGetCode();
//if (code == 200) { saveResponse(msg); }
//if (code >= 400) { mFailCode = code; }
//SipDialog::msgState nextDialogState = sipMessage2DialogState(msg);
switch (mDialogType) {
case SIPDTRegister:
case SIPDTUnregister:
LOG(ERR) << "REGISTER transaction received unexpected message:"<<msg;
break;
case SIPDTMOUssd:
LOG(DEBUG);
if (code == 0 && msg->isBYE()) { // It is a SIP Request. Switch based on the method.
// Grab any xml ussd response from the BYE message.
handleUssdBye(msg);
}
goto otherwise;
case SIPDTMOC: // This is a MOC transaction.
case SIPDTMTC: // This is a MTC transaction. Could be an inbound handover.
LOG(DEBUG);
otherwise:
if (code == 0) { // It is a SIP Request. Switch based on the method.
if (msg->isBYE()) {
SipMTBye(msg);
} else if (msg->isCANCEL()) {
// This is an error since we have already passed the ACK stage, but lets cancel the dialog anyway.
SipMTCancel(msg);
} else {
// Not expecting any others. Must send 405 error.
LOG(ALERT)<<"SIP Message ignored:"<<msg; // TEMPORARY: Make this show up.
LOG(WARNING)<<"SIP Message ignored:"<<msg;
SipMessageReply oops(msg,405,string("Method Not Allowed"),this);
sipWrite(&oops);
}
} else {
// This should have matched a Transaction somewhere.
// We cant send an error back for an unrecognized response or we get in an infinite loop.
LOG(ALERT) << "SIP response not handled:"<<msg;
}
break;
case SIPDTMOSMS:
case SIPDTMTSMS:
LOG(ERR) << "MESSAGE transaction received unexpected message:"<<msg;
break;
default:
assert(0);
}
dialogChangeState(msg);
delete msg;
}
// This is only called after the dialog has already been removed from the active dialogs,
// so we dont have to check the dialog state, all we have to check is there is nothing pointing to it
// that would cause a crash if genuinely deleted.
bool SipDialog::dgIsDeletable() const
{
ScopedLock lock(mDialogLock,__FILE__,__LINE__);
switch (mDialogType) {
case SIPDTMOC:
case SIPDTMTC:
case SIPDTMOSMS:
case SIPDTMTSMS:
case SIPDTMOUssd:
return gNewTransactionTable.ttIsDialogReleased(this->mTranId);
// We never expire the dialog associated with REGISTER.
case SIPDTRegister:
case SIPDTUnregister:
case SIPDTUndefined:
return false; // We never delete the Register dialog.
default:
assert(0);
}
}
// Called periodicially to check for SIP timer expiration.
bool SipDialog::dialogPeriodicService()
{
// Take care. This is a potential deadlock if somone tries to add a locked SipDialog into the DialogMap,
// because the kicker code locks the whole DialogMap against modification.
ScopedLock lock(mDialogLock,__FILE__,__LINE__);
// Now we use TransactionUsers for client transactions, so this code handles only server transactions.
// The in-dialog server transactions are trivial - the transaction-layer simply resends the final
// response each time the request is received.
switch (mDialogType) {
case SIPDTUndefined:
case SIPDTRegister:
case SIPDTUnregister:
// FIXME: I dont think we delete these, ever.
break;
case SIPDTMTSMS:
case SIPDTMTC:
return mtPeriodicService();
break;
case SIPDTMOSMS:
case SIPDTMOC:
case SIPDTMOUssd:
return moPeriodicService();
break;
//default: break;
}
return false;
}
const char *DialogState::msgStateString(DialogState::msgState dstate)
{
switch (dstate) {
case DialogState::dialogUndefined: return "undefined";
case DialogState::dialogStarted: return "Started";
case DialogState::dialogProceeding: return "Proceeding";
case DialogState::dialogRinging: return "Ringing";
case DialogState::dialogActive: return "Active";
case DialogState::dialogBye: return "Bye";
case DialogState::dialogFail: return "Fail";
case DialogState::dialogDtmf: return "DTMF";
};
return "unknown_DialogState";
}
string SipDialog::dialogText(bool verbose) const
{
std::ostringstream ss;
ss << " SipDialog("<<LOGVARM(mTranId) ;
ss << LOGVAR2("state",getDialogState()) <<LOGVARM(mPrevDialogState);
//ss << LOGVAR2("fifo",mDownlinkFifo.size());
sdbText(ss,verbose);
// The C++ virtual inheritance is so broken we cant use it. Gag me.
switch (mDialogType) {
case SIPDTMTC: case SIPDTMTSMS:
ss << mttlText();
break;
case SIPDTMOC: case SIPDTMOSMS: case SIPDTMOUssd:
ss << motlText();
break;
default: ss << "."; break;
}
ss <<")";
return ss.str();
}
std::ostream& operator<<(std::ostream& os, const SipDialog*dg) {
if (dg) os << dg->dialogText(); else os << "(null SipDialog)";
return os;
}
std::ostream& operator<<(std::ostream& os, const SipDialog&dg) { os << dg.dialogText(); return os; } // stupid language
std::ostream& operator<<(std::ostream& os, const SipDialogRef&dr) {
SipDialog *dg = dr.self();
if (dg) os << dg->dialogText(); else os << "(null SipDialog)";
return os;
}
std::ostream& operator<<(std::ostream& os, const DialogState::msgState dstate)
{
os << DialogState::msgStateString(dstate);
return os;
}
std::ostream& operator<<(std::ostream& os, const DialogMessage*dmsg)
{
if (dmsg) {
os <<"DialogMessage("<<LOGVAR2("MsgState",DialogState::msgStateString(dmsg->mMsgState)) <<LOGVAR2("StatusCode",dmsg->mSipStatusCode)<<")";
} else {
os << "(null DialogMessage)";
}
return os;
}
std::ostream& operator<<(std::ostream& os, const DialogMessage&dmsg) { os << &dmsg; return os; } // stupid language
}; // namespace
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