mirror of
https://github.com/PentHertz/OpenBTS.git
synced 2026-05-10 12:00:03 +00:00
Oh kay. Lots of patches trying to catch up to mainline:
r4232: Major change in hos entries are removed from a TransactionTable. >From now on, the only place were entries are actually removed is in TransactionTable::innerRemove. Anywhere else, the remove methods just set a flag to tag the entry for later removal be innerRemove. This allows us to survive situations where we try to touch transactions that have been removed already. We still log at the ERR level, but no more segfaults. Updated all of the TransactionTable "find" methods to not return pointers to dead or removed tranactions. Updated find-by-channel search to return the transaction entry with the highest transaction number, which fixes a bug that sometimes picked up the wrong transaction records during EA TCH assignment. r4253: New exception class for when someone tries to use a "removed" transaction. r4254: Updated copyright notice. r4265: Unlock TransactionEntry::mLock while blocked on SIP message FIFOs in SIPEngine. This does wonders to reduce lock contention and make everything more snappy. Use Mutex::tryLock in TransactionEntry::dead and if lock is held, assume that it is not dead. This also does a lot to reduce lock contention. r4294: Change Um congestion response back to SIP 503. r4295: When deleting an entry that has failed to respond to paging, send a SIP 480 response so the switch knows that transaction is dead. r4412: Fixed bug that was causing SIPInterface to crash when the IMSI cannot be extracted from the To: header. git-svn-id: http://wush.net/svn/range/software/public/openbts/trunk@4497 19bc5d8c-e614-43d4-8b26-e1612bc8e597
This commit is contained in:
Kurtis Heimerl
parent
9536ac91dd
commit
768815e07f
11 changed files with 876 additions and 302 deletions
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@ -40,11 +40,13 @@
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#include <GSMConfig.h>
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#include <ControlCommon.h>
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#include <GSMCommon.h>
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#include <GSMLogicalChannel.h>
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#include "SIPInterface.h"
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#include "SIPUtility.h"
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#include "SIPMessage.h"
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#include "SIPEngine.h"
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#include "TransactionTable.h"
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#undef WARNING
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@ -245,6 +247,44 @@ void SIPEngine::user( const char * wCallID, const char * IMSI, const char *origI
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}
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void SIPEngine::writePrivateHeaders(osip_message_t *msg, const GSM::LogicalChannel *chan)
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{
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// P-PHY-Info
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// This is a non-standard private header in OpenBTS.
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// TA=<timing advance> TE=<TA error> UpRSSI=<uplink RSSI> TxPwr=<MS tx power> DnRSSIdBm=<downlink RSSI>
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// Get the values
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#if 0
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if (chan) {
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char phy_info[200];
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sprintf(phy_info,"OpenBTS; TA=%d TE=%f UpRSSI=%f TxPwr=%d DnRSSIdBm=%d",
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chan->actualMSTiming(), chan->timingError(),
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chan->RSSI(), chan->actualMSPower(),
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chan->measurementResults().RXLEV_FULL_SERVING_CELL_dBm());
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osip_message_set_header(msg,"P-PHY-Info",phy_info);
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}
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#endif
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// P-Access-Network-Info
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// See 3GPP 24.229 7.2.
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char cgi_3gpp[50];
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sprintf(cgi_3gpp,"3GPP-GERAN; cgi-3gpp=%s%s%04x%04x",
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gConfig.getStr("GSM.Identity.MCC").c_str(),gConfig.getStr("GSM.Identity.MNC").c_str(),
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(unsigned)gConfig.getNum("GSM.Identity.LAC"),(unsigned)gConfig.getNum("GSM.Identity.CI"));
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osip_message_set_header(msg,"P-Access-Network-Info",cgi_3gpp);
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// P-Preferred-Identity
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// See RFC-3325.
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char pref_id[50];
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sprintf(pref_id,"<sip:%s@%s>",
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mSIPUsername.c_str(),
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gConfig.getStr("SIP.Proxy.Speech").c_str());
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osip_message_set_header(msg,"P-Preferred-Identity",pref_id);
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// FIXME -- Use the subscriber registry to look up the E.164
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// and make a second P-Preferred-Identity header.
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}
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bool SIPEngine::Register( Method wMethod )
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{
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LOG(INFO) << "user " << mSIPUsername << " state " << mState << " " << wMethod << " callID " << mCallID;
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@ -355,7 +395,7 @@ const char* geoprivTemplate =
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"</tuple>\n"
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"</presence>\n";
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SIPState SIPEngine::SOSSendINVITE(short wRtp_port, unsigned wCodec)
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SIPState SIPEngine::SOSSendINVITE(short wRtp_port, unsigned wCodec, const GSM::LogicalChannel *chan)
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{
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LOG(INFO) << "user " << mSIPUsername << " state " << mState;
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// Before start, need to add mCallID
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@ -390,26 +430,7 @@ SIPState SIPEngine::SOSSendINVITE(short wRtp_port, unsigned wCodec)
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mMyTag.c_str(), mViaBranch.c_str(), mCallID.c_str(), mCSeq, mCodec);
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}
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// Add IMS emergency call headers with osip_message_set_header.
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// P-Access-Network-Info
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// See 3GPP 24.229 7.2.
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char cgi_3gpp[50];
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sprintf(cgi_3gpp,"3GPP-GERAN; cgi-3gpp=%s%s%04x%04x",
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gConfig.getStr("GSM.Identity.MCC").c_str(),gConfig.getStr("GSM.Identity.MNC").c_str(),
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(unsigned)gConfig.getNum("GSM.Identity.LAC"),(unsigned)gConfig.getNum("GSM.Identity.CI"));
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osip_message_set_header(invite,"P-Access-Network-Info",cgi_3gpp);
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// P-Preferred-Identity
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// See RFC-3325.
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char pref_id[50];
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sprintf(pref_id,"<sip:%s@%s>",
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mSIPUsername.c_str(),
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gConfig.getStr("Control.Emergency.GatewaySwitch").c_str());
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osip_message_set_header(invite,"P-Preferred-Identity",pref_id);
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// FIXME -- Use the subscriber registry to look up the E.164
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// and make a second P-Preferred-Identity header.
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writePrivateHeaders(invite,chan);
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// Add RFC-4119 geolocation XML to content area, if available.
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if (gConfig.defines("Control.Emergency.Geolocation")) {
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@ -434,7 +455,8 @@ SIPState SIPEngine::SOSSendINVITE(short wRtp_port, unsigned wCodec)
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SIPState SIPEngine::MOCSendINVITE( const char * wCalledUsername,
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const char * wCalledDomain , short wRtp_port, unsigned wCodec)
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const char * wCalledDomain , short wRtp_port, unsigned wCodec,
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const GSM::LogicalChannel *chan)
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{
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LOG(INFO) << "user " << mSIPUsername << " state " << mState;
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// Before start, need to add mCallID
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@ -461,14 +483,7 @@ SIPState SIPEngine::MOCSendINVITE( const char * wCalledUsername,
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mSIPPort, mSIPIP.c_str(), mProxyIP.c_str(),
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mMyTag.c_str(), mViaBranch.c_str(), mCallID.c_str(), mCSeq, mCodec);
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// P-Access-Network-Info
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// See 3GPP 24.229 7.2.
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char cgi_3gpp[50];
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sprintf(cgi_3gpp,"3GPP-GERAN; cgi-3gpp=%s%s%04x%04x",
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gConfig.getStr("GSM.Identity.MCC").c_str(),gConfig.getStr("GSM.Identity.MNC").c_str(),
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(unsigned)gConfig.getNum("GSM.Identity.LAC"),(unsigned)gConfig.getNum("GSM.Identity.CI"));
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osip_message_set_header(invite,"P-Access-Network-Info",cgi_3gpp);
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writePrivateHeaders(invite,chan);
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// Send Invite.
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gSIPInterface.write(&mProxyAddr,invite);
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@ -487,7 +502,7 @@ SIPState SIPEngine::MOCResendINVITE()
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return mState;
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}
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SIPState SIPEngine::MOCWaitForOK()
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SIPState SIPEngine::MOCCheckForOK(Mutex *lock)
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{
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LOG(INFO) << "user " << mSIPUsername << " state " << mState;
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@ -496,7 +511,7 @@ SIPState SIPEngine::MOCWaitForOK()
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// Read off the fifo. if time out will
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// clean up and return false.
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try {
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msg = gSIPInterface.read(mCallID, gConfig.getNum("SIP.Timer.A"));
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msg = gSIPInterface.read(mCallID, gConfig.getNum("SIP.Timer.A"),lock);
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}
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catch (SIPTimeout& e) {
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LOG(DEBUG) << "timeout";
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@ -656,12 +671,12 @@ SIPState SIPEngine::MODResendERROR(bool cancel)
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/* there shouldn't be any more communications on this fifo, but we might
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get a 487 RequestTerminated. We only need to respond and move on -kurtis */
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SIPState SIPEngine::MODWaitFor487()
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SIPState SIPEngine::MODWaitFor487(Mutex *lock)
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{
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LOG(INFO) << "user " << mSIPUsername << " state " << mState;
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osip_message_t * msg;
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try {
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msg = gSIPInterface.read(mCallID, gConfig.getNum("SIP.Timer.E"));
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msg = gSIPInterface.read(mCallID, gConfig.getNum("SIP.Timer.E"), lock);
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}
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catch (SIPTimeout& e) {
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LOG(NOTICE) << "487 Timeout";
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@ -686,14 +701,14 @@ SIPState SIPEngine::MODWaitFor487()
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}
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}
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SIPState SIPEngine::MODWaitForBYEOK()
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SIPState SIPEngine::MODWaitForBYEOK(Mutex *lock)
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{
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LOG(INFO) << "user " << mSIPUsername << " state " << mState;
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bool responded = false;
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Timeval timeout(gConfig.getNum("SIP.Timer.F"));
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while (!timeout.passed()) {
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try {
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osip_message_t * ok = gSIPInterface.read(mCallID, gConfig.getNum("SIP.Timer.E"));
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osip_message_t * ok = gSIPInterface.read(mCallID, gConfig.getNum("SIP.Timer.E"),lock);
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responded = true;
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unsigned code = ok->status_code;
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saveResponse(ok);
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@ -716,14 +731,14 @@ SIPState SIPEngine::MODWaitForBYEOK()
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return mState;
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}
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SIPState SIPEngine::MODWaitForCANCELOK()
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SIPState SIPEngine::MODWaitForCANCELOK(Mutex *lock)
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{
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LOG(INFO) << "user " << mSIPUsername << " state " << mState;
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bool responded = false;
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Timeval timeout(gConfig.getNum("SIP.Timer.F"));
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while (!timeout.passed()) {
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try {
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osip_message_t * ok = gSIPInterface.read(mCallID, gConfig.getNum("SIP.Timer.E"));
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osip_message_t * ok = gSIPInterface.read(mCallID, gConfig.getNum("SIP.Timer.E"),lock);
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responded = true;
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unsigned code = ok->status_code;
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saveResponse(ok);
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@ -746,14 +761,65 @@ SIPState SIPEngine::MODWaitForCANCELOK()
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return mState;
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}
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SIPState SIPEngine::MODWaitForERRORACK(bool cancel)
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static bool containsResponse(vector<unsigned> *validResponses, unsigned code)
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{
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for (int i = 0; i < validResponses->size(); i++) {
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if (validResponses->at(i) == code)
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return true;
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}
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return false;
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}
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SIPState SIPEngine::MODWaitForResponse(vector<unsigned> *validResponses, Mutex *lock)
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{
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LOG(INFO) << "user " << mSIPUsername << " state " << mState;
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assert(validResponses);
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bool responded = false;
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Timeval timeout(gConfig.getNum("SIP.Timer.F"));
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while (!timeout.passed()) {
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try {
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osip_message_t * resp = gSIPInterface.read(mCallID, gConfig.getNum("SIP.Timer.E"),lock);
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responded = true;
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unsigned code = resp->status_code;
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if (code==200) {
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saveResponse(resp);
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mState = Canceled;
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}
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if (code==487) {
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osip_message_t* ack = sip_ack( mRemoteDomain.c_str(),
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mRemoteUsername.c_str(),
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mSIPUsername.c_str(),
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mSIPPort, mSIPIP.c_str(), mProxyIP.c_str(),
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mMyToFromHeader, mRemoteToFromHeader,
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mViaBranch.c_str(), mCallIDHeader, mCSeq);
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gSIPInterface.write(&mProxyAddr,ack);
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osip_message_free(ack);
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}
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osip_message_free(resp);
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if (!containsResponse(validResponses, code)) {
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LOG(WARNING) << "unexpected " << code << " response to CANCEL, from proxy " << mProxyIP << ":" << mProxyPort << ". Assuming other end has cleared";
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}
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break;
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}
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catch (SIPTimeout& e) {
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LOG(NOTICE) << "response timeout, resending CANCEL";
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MODResendCANCEL();
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}
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}
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if (!responded) { LOG(ALERT) << "lost contact with proxy " << mProxyIP << ":" << mProxyPort; }
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return mState;
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}
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SIPState SIPEngine::MODWaitForERRORACK(bool cancel, Mutex *lock)
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{
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LOG(INFO) << "user " << mSIPUsername << " state " << mState;
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bool responded = false;
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Timeval timeout(gConfig.getNum("SIP.Timer.F"));
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while (!timeout.passed()) {
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try {
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osip_message_t * ack = gSIPInterface.read(mCallID, gConfig.getNum("SIP.Timer.E"));
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osip_message_t * ack = gSIPInterface.read(mCallID, gConfig.getNum("SIP.Timer.E"),lock);
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responded = true;
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saveResponse(ack);
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if ((NULL == ack->sip_method) || !strncmp(ack->sip_method,"ACK", 4)) {
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@ -881,7 +947,7 @@ SIPState SIPEngine::MTCSendRinging()
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SIPState SIPEngine::MTCSendOK( short wRTPPort, unsigned wCodec )
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SIPState SIPEngine::MTCSendOK( short wRTPPort, unsigned wCodec, const GSM::LogicalChannel *chan)
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{
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LOG(INFO) << "user " << mSIPUsername << " state " << mState;
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assert(mINVITE);
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@ -891,13 +957,14 @@ SIPState SIPEngine::MTCSendOK( short wRTPPort, unsigned wCodec )
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// Form ack from invite and new parameters.
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osip_message_t * okay = sip_okay_sdp(mINVITE, mSIPUsername.c_str(),
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mSIPIP.c_str(), mSIPPort, mRTPPort, mCodec);
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writePrivateHeaders(okay,chan);
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gSIPInterface.write(&mProxyAddr,okay);
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osip_message_free(okay);
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mState=Connecting;
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return mState;
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}
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SIPState SIPEngine::MTCWaitForACK()
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SIPState SIPEngine::MTCCheckForACK(Mutex *lock)
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{
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// wait for ack,set this to timeout of
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// of call channel. If want a longer timeout
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@ -908,7 +975,7 @@ SIPState SIPEngine::MTCWaitForACK()
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// FIXME -- This is supposed to retransmit BYE on timer I.
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try {
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ack = gSIPInterface.read(mCallID, gConfig.getNum("SIP.Timer.H"));
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ack = gSIPInterface.read(mCallID, gConfig.getNum("SIP.Timer.H"), lock);
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}
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catch (SIPTimeout& e) {
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LOG(NOTICE) << "timeout";
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@ -1126,7 +1193,8 @@ int SIPEngine::rxFrame(unsigned char* frame)
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SIPState SIPEngine::MOSMSSendMESSAGE(const char * wCalledUsername,
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const char * wCalledDomain , const char *messageText, const char *contentType)
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const char * wCalledDomain , const char *messageText, const char *contentType,
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const GSM::LogicalChannel *chan)
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{
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LOG(DEBUG) << "mState=" << mState;
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LOG(INFO) << "SIP send to " << wCalledUsername << "@" << wCalledDomain << " MESSAGE " << messageText;
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@ -1148,7 +1216,9 @@ SIPState SIPEngine::MOSMSSendMESSAGE(const char * wCalledUsername,
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mSIPPort, mSIPIP.c_str(), mProxyIP.c_str(),
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mMyTag.c_str(), mViaBranch.c_str(), mCallID.c_str(), mCSeq,
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messageText, contentType);
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writePrivateHeaders(message,chan);
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// Send Invite to the SIP proxy.
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gSIPInterface.write(&mProxyAddr,message);
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saveINVITE(message,true);
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@ -1158,7 +1228,7 @@ SIPState SIPEngine::MOSMSSendMESSAGE(const char * wCalledUsername,
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};
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SIPState SIPEngine::MOSMSWaitForSubmit()
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SIPState SIPEngine::MOSMSWaitForSubmit(Mutex *lock)
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{
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LOG(INFO) << "user " << mSIPUsername << " state " << mState;
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@ -1171,7 +1241,7 @@ SIPState SIPEngine::MOSMSWaitForSubmit()
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try {
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// SIPInterface::read will throw SIPTIimeout if it times out.
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// It should not return NULL.
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ok = gSIPInterface.read(mCallID, gConfig.getNum("SIP.Timer.A"));
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ok = gSIPInterface.read(mCallID, gConfig.getNum("SIP.Timer.A"),lock);
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}
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catch (SIPTimeout& e) {
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if (!recv_trying){
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@ -1215,7 +1285,7 @@ SIPState SIPEngine::MOSMSWaitForSubmit()
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}
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SIPState SIPEngine::MTSMSSendOK()
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SIPState SIPEngine::MTSMSSendOK(const GSM::LogicalChannel *chan)
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{
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LOG(INFO) << "user " << mSIPUsername << " state " << mState;
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// If this operation was initiated from the CLI, there was no INVITE.
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@ -1227,6 +1297,7 @@ SIPState SIPEngine::MTSMSSendOK()
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// Form ack from invite and new parameters.
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osip_message_t * okay = sip_okay(mINVITE, mSIPUsername.c_str(),
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mSIPIP.c_str(), mSIPPort);
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writePrivateHeaders(okay,chan);
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gSIPInterface.write(&mProxyAddr,okay);
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osip_message_free(okay);
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mState=Cleared;
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@ -1235,7 +1306,7 @@ SIPState SIPEngine::MTSMSSendOK()
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bool SIPEngine::sendINFOAndWaitForOK(unsigned wInfo)
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bool SIPEngine::sendINFOAndWaitForOK(unsigned wInfo, Mutex *lock)
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{
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LOG(INFO) << "user " << mSIPUsername << " state " << mState;
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@ -1248,23 +1319,32 @@ bool SIPEngine::sendINFOAndWaitForOK(unsigned wInfo)
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mSIPPort, mSIPIP.c_str(), mProxyIP.c_str(),
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mMyTag.c_str(), mViaBranch.c_str(), mCallIDHeader, mCSeq);
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gSIPInterface.write(&mProxyAddr,info);
|
||||
osip_message_free(info);
|
||||
|
||||
try {
|
||||
// This will timeout on failure. It will not return NULL.
|
||||
osip_message_t *msg = gSIPInterface.read(mCallID, gConfig.getNum("SIP.Timer.A"));
|
||||
LOG(DEBUG) << "received status " << msg->status_code << " " << msg->reason_phrase;
|
||||
bool retVal = (msg->status_code==200);
|
||||
osip_message_free(msg);
|
||||
if (!retVal) LOG(CRIT) << "DTMF RFC-2967 failed.";
|
||||
return retVal;
|
||||
Timeval timeout(gConfig.getNum("SIP.Timer.F"));
|
||||
osip_message_t *ok = NULL;
|
||||
while (!timeout.passed()) {
|
||||
try {
|
||||
// This will timeout on failure. It will not return NULL.
|
||||
ok = gSIPInterface.read(mCallID, gConfig.getNum("SIP.Timer.E"), lock);
|
||||
LOG(DEBUG) << "received status " << ok->status_code << " " << ok->reason_phrase;
|
||||
}
|
||||
catch (SIPTimeout& e) {
|
||||
LOG(NOTICE) << "SIP RFC-2967 INFO packet to " << mProxyIP << ":" << mProxyPort << " timedout; resending";
|
||||
gSIPInterface.write(&mProxyAddr,info);
|
||||
continue;
|
||||
}
|
||||
}
|
||||
catch (SIPTimeout& e) {
|
||||
LOG(NOTICE) << "timeout";
|
||||
osip_message_free(info);
|
||||
if (!ok) {
|
||||
LOG(ALERT) << "SIP RFC-2967 INFO timed out; is the proxy at " << mProxyIP << ":" << mProxyPort << " OK?";
|
||||
return false;
|
||||
}
|
||||
|
||||
};
|
||||
LOG(DEBUG) << "received status " << ok->status_code << " " << ok->reason_phrase;
|
||||
bool retVal = (ok->status_code==200);
|
||||
osip_message_free(ok);
|
||||
if (!retVal) LOG(WARNING) << "SIP RFC-2967 INFO failed on server " << mProxyIP << ":" << mProxyPort << " OK?";
|
||||
return retVal;
|
||||
}
|
||||
|
||||
/* reinvite stuff */
|
||||
/* return true if this is the same invite as the one we have stored */
|
||||
|
|
|
|||
Loading…
Add table
Add a link
Reference in a new issue