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From: OpenSIPStack F. <ope...@op...> - 2008-06-19 03:35:00
|
Yes, it is properly closed. For some reason the copy and paste didn't correctly show what I had. here is what I have: <siptrunk trunk-name="les.net" route-set="did.les.net" sip-domain="did.les.net" expires="3600"> <trunk-accounts> <account user-name="xxxx" auth-user-name="xxxx" auth-password="xxxx" inbound-route="sip:100@domain" expires="3600"> </trunk-accounts> </siptrunk> Please help :( Im starting the program with : sudo ./opensbc -xc Would that cause a problem? |
|
From: OpenSIPStack F. <ope...@op...> - 2008-06-19 01:15:44
|
my guess is you did not enclose your xml entry with <root></root>. Perhaps this post would help. http://www.opensourcesip.org:8080/clearspacex/message/8040#8040 Joegen > {quote:title=illizit wrote:}{quote} > Hello! > > I am having issues registering a trunk with my voip provider. I am using ngrep and cannot see any attempts to register at all. > > > I have trunk ports enabled in the general config. > > > I have this in my sip trunk configuration: > > > <siptrunk trunk-name="les.net" > route-set="did.les.net" > sip-domain="did.les.net" > expires="3600"> > <trunk-accounts> > <account user-name="xxxx" > auth-user-name="xxxx" > auth-password="xxxxx" > inbound-route="sip:10...@my..." > expires="3600" /> > > > > > > Please help. > > > Thanks! |
|
From: Joegen E. B. <joe...@gm...> - 2008-06-19 01:02:26
|
Hi Gustavo, I've been tracking this issue in assembla. Feel free to create an account in the tracker so you could comment on the status of the leak. http://www.assembla.com/spaces/opensbc/tickets/22 Joegen Joegen E. Baclor wrote: > Right! I've found m_MinSE is not getting deleted in > SIPMessage::CleanUp(). Patched this in CVS. > > Joegen > > Gustavo Curetti wrote: > >> Hi Joegen >> >> The modification doesn't solve the memory issues. I continue searching >> for the memory leak. >> >> A new case is attached and this one appear too when debugging with >> Microsot Visual. >> >> Originally, i sent the attached Invite every 250 ms and I set the >> timer B and H in 20 ms: >> >> #define SIP_TIMER_B 20 >> >> #define SIP_TIMER_H 20 >> >> Then I change the code >> of B2BUserAgent::Registrar::ProcessUpperRegKeepAlive() >> for making easier to replicate the leak: >> >> >> void B2BUserAgent::Registrar::ProcessUpperRegKeepAlive() >> { >> while( !m_UpperRegSync.Wait( 250 ) ) >> { >> OString testRequest = >> "INVITE sip:5435155555@192.168.0.5:5060 SIP/2.0\r\nContact: >> <sip:4284623@192.168.0.10:5060>\r\nCSeq: 101 INVITE\r\nFrom: >> <sip:4284623@192.168.0.10>;tag=5A3745C-2418\r\nTo: >> <sip:55555555@192.168.0.206>\r\nVia: SIP/2.0/UDP >> 192.168.0.206:5060;branch=z9hG4bK63028de3a6b7743a\r\nVia: SIP/2.0/UDP >> 192.168.0.10:5060\r\nRecord-Route: >> <sip:192.168.0.206:5060;lr>\r\nAllow: INVITE, OPTIONS, BYE, CANCEL, >> ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO\r\nUser-Agent: >> Cisco-SIPGateway/IOS-12.x\r\nCall-Id: >> 3738EB25-278011DD-B92F90A6-C0EF6BE3@192.168.0.10\r\nMax-Forwards >> <mailto:3738EB25-278011DD-B92F90A6-C0EF6BE3@192.168.0.10%5Cr%5CnMax-Forwards>: >> 6\r\nExpires: 180\r\nContent-Length: 235\r\ndate: Thu, 22 May 2008 >> 21:52:32 GMT\r\nsupported: timer\r\nmin-se: 1800\r\ncisco-guid: >> 926237238-662704605-3106705574-3236916195\r\nremote-party-id: >> <sip:4284623@192.168.0.10>;party=calling;screen=no;privacy=off\r\ntimestamp: >> 1211493152\r\nallow-events: telephone-event\r\ncontent-type: >> application/sdp\r\n\r\nv=0\r\no=CiscoSystemsSIP-GW-UserAgent 7402 717 >> IN IP4 192.168.0.10\r\ns=SIP Call\r\nc=IN IP4 192.168.0.10\r\nt=0 >> 0\r\nm=audio 19298 RTP/AVP 0 19\r\nc=IN IP4 192.168.0.10\r\na=rtpmap:0 >> PCMU/8000\r\na=rtpmap:19 CN/8000\r\na=ptime:20"; >> testRequest = ParserTools::LineFeedSanityCheck( testRequest ); >> SIPMessage * msg = new SIPMessage( testRequest ); >> >> OString addrStr = "192.168.0.147"; >> OString portStr = "10000"; >> SIPHeader rcvAddr( "RCVADDR", addrStr ); >> SIPHeader rcvPort( "RCVPORT", portStr ); >> SIPHeader rcvTran( "RCVTRAN", "udp" ); >> >> msg->AddInternalHeader( rcvAddr ); >> msg->AddInternalHeader( rcvPort ); >> msg->AddInternalHeader( rcvTran ); >> msg->SetInterfaceAddress( "192.168.0.202" ); >> msg->SetInterfacePort( 5070 ); >> >> OStringStream traceStream; >> >> traceStream << "<<< " >> << msg->GetStartLine() << " " >> << " SRC: " << addrStr << ":" << portStr << ":UDP" >> << " enc=" << msg->IsEncrypted() >> << " bytes=1103"; >> >> OStringStream strPacket; >> strPacket << *msg; >> COMPOUND_LOG_CONTEXT( LogInfo(), msg->GetCallId(), >> traceStream.str(), LogDebugHigh(), strPacket ); >> >> SIPTransport::NotifyRead( traceStream.str() ); >> >> if( msg->IsInvite() ) >> { >> SIPMessage * trying = new SIPMessage(); >> msg->CreateResponse( *trying, SIPMessage::Code100_Trying ); >> Via via; >> msg->GetViaAt(0, via ); >> if( via.IsBehindNAT() ) >> { >> SIPURI srcURI; >> srcURI.SetHost(addrStr); >> srcURI.SetPort(portStr); >> trying->SetSendAddress(srcURI); >> } >> if( msg->IsEncrypted() ) >> trying->SetEncryption( TRUE ); >> GetTransportManager()->ProcessOutbound( trying ); >> } >> >> GetTransportManager()->OnTransportEvent( >> new SIPTransportEvent( >> msg, >> SIPTransportEvent::UDPPacketArrival >> ) ); >> >> /*///process the keep alives here >> for( PINDEX i = 0; i < GetRegistrationDB().GetSize(); i++ ) >> { >> SIPMessage reg; >> if( GetRegistrationDB().GetRegistration( i, reg ) ) >> { >> if( reg.HasInternalHeader( "upper-reg" ) ) >> { >> /// this is an upper reg, send a keep-alive >> /// Check the last via if its from a private IP >> Via via; >> if( reg.GetViaAt( reg.GetViaSize() - 1, via ) ) >> { >> if( via.IsBehindNAT() ) >> { >> SIPURI target; >> target.SetHost( via.GetReceiveAddress().AsSTLString() ); >> target.SetPort( via.GetRPort() ); >> SIPMessage keepAlive; >> RequestLine requestLine; >> requestLine.SetMethod( "KEEP-ALIVE" ); >> requestLine.SetRequestURI( target ); >> keepAlive.SetStartLine( requestLine ); >> GetUserAgent().TransportWrite( keepAlive ); >> } >> } >> } >> } >> }*/ >> } >> } >> >> >> The OpenSBC is in "Proxy Only Mode" and the configuration is in >> "OpenSBC.reg" (attached). >> >> I compile the OpenSBC in Microsoft Visual C++ 2005 obtaining the exe >> attached. >> >> Any idea? >> >> Thanks for your help. >> >> Gustavo >> >> >> >> ------------------------------------------------------------------------ >> >> >>> Date: Thu, 5 Jun 2008 14:18:35 +0800 >>> To: cur...@gm... >>> Subject: Re: [OpenSIPStack] FW: Memory Leak in Proxy and Full Mode >>> From: joe...@gm... >>> >>> Hi Gustavo, >>> >>> Yes, i'm able to replicate it. For some reason, the code I #ifdefed in >>> AppendVia() below is causing it. Let me know if it solves your issues. >>> I've tried looking at what its doing but nothing is evident as to >>> >> why it >> >>> would leak. If you find something, let me know >>> >>> Joegen >>> >>> BOOL SIPMessage::AppendVia( >>> const Via & header >>> ) >>> { >>> GlobalLock(); >>> >>> ParseViaList(); >>> >>> if( m_ViaList == NULL ) >>> { >>> m_ViaList = new Via::Collection(); >>> m_ViaList->Append( new Via( header ) ); >>> }else >>> { >>> >>> #if 0 // For some reason, this sanity check is leaking mem >>> /// sanity check >>> if( m_ViaList->GetSize() > 0 ) >>> { >>> Via & topVia = (*m_ViaList)[0]; >>> >>> SIPURI topViaURI = topVia.GetURI(); >>> >>> SIPURI newURI = header.GetURI(); >>> >>> if( SIPTransport::IsTheSameAddress( topViaURI, newURI, TRUE ) ) >>> return FALSE; >>> } >>> #endif >>> >>> ///we append it on top >>> Via::Collection oldViaList = *m_ViaList; >>> oldViaList.MakeUnique(); >>> delete m_ViaList; >>> >>> m_ViaList = new Via::Collection(); >>> m_ViaList->Append( new Via( header ) ); >>> >>> for( PINDEX i = 0; i < oldViaList.GetSize(); i++ ) >>> m_ViaList->Append( new Via( oldViaList[i] ) ); >>> >>> m_ViaList->MakeUnique(); >>> } >>> >>> >>> return TRUE; >>> } >>> >>> >>> Gustavo Curetti wrote: >>> >>>> Hi Joegen >>>> >>>> The destructor is called. The problem seem to be the headers like >>>> >> Via, >> >>>> RecordRoute, Contact, Allow, Supported (List headers). >>>> >>>> The leak is very easy to reproduce. I change the code of >>>> B2BUserAgent::Registrar::ProcessUpperRegKeepAlive: >>>> >>>> >>>> void B2BUserAgent::Registrar::ProcessUpperRegKeepAlive() >>>> { >>>> while( !m_UpperRegSync.Wait( 10 ) ) >>>> { >>>> >>>> SIPMessage * msg = new SIPMessage(); >>>> >>>> msg->AppendVia(Via("SIP/2.0/UDP 192.168.0.10:5060")); >>>> msg->AppendVia(Via("SIP/2.0/UDP >>>> >>>> >> 192.168.0.206:5060;branch=z9hG4bK440fdc3e04de9d10;rport=5060;received=192.168.0.206")); >> >>>> msg->AppendRecordRoute(RouteURI("<sip:192.168.0.206:5060;lr>")); >>>> msg->AppendContact(ContactURI("<sip:5435155555@192.168.0.5>")); >>>> msg->AppendAllow(Allow("INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, >>>> SUBSCRIBE, NOTIFY")); >>>> msg->AppendSupported(Supported("replaces")); >>>> >>>> delete msg; >>>> } >>>> } >>>> >>>> >>>> I compile the OpenSBC in Microsoft Visual C++ 2005 obtaining the exe >>>> attached. >>>> I execute "OpenSBC Debug" in Windows 2003 or Windows 2000 and the >>>> >> leak >> >>>> is there. >>>> >>>> I put traces and SIPMessage::~SIPMessage(), SIPMessage::Cleanup() are >>>> called. >>>> >>>> Any idea? >>>> >>>> Thanks for your help. >>>> >>>> Gustavo >>>> >>>> >>>> >> ------------------------------------------------------------------------ >> >>>>> Date: Mon, 2 Jun 2008 10:19:08 +0800 >>>>> To: cur...@gm...; >>>>> >> ope...@li... >> >>>>> From: joe...@gm... >>>>> Subject: Re: [OpenSIPStack] FW: Memory Leak in Proxy and Full Mode >>>>> >>>>> Try putting a trace before and after the "delete" statement to >>>>> >> be sure >> >>>>> that its getting called. From what i see, assuming that SIPMessage >>>>> destructor performs proper cleanup, that a leak here can only be >>>>> >> caused >> >>>>> by an exception occurring in the SIPMessage constructor; >>>>> >>>>> Gustavo Curetti wrote: >>>>> >>>>>> Hi Joegen: >>>>>> >>>>>> The memory leak is hard to find. I send the 200 OK repeatedly but >>>>>> >>>> the memory leak doesn't appear if i'm debugging with Microsot Visual >>>> 2005, but the leak appears if i'm running the OpenSBC like a service. >>>> >>>>>> If I comment: >>>>>> >>>>>> >>>>>> void SIPStack::EnqueueTransportWriteEvent( >>>>>> const SIPMessage & message, >>>>>> SIPTransportEvent::Type eventType >>>>>> ) >>>>>> { >>>>>> if( m_IsTerminating ) >>>>>> return; >>>>>> >>>>>> /*SIPMessage * msg = new SIPMessage(message); >>>>>> >>>>>> msg->SetInternalHeader( "TRN-ID", >>>>>> >> msg->GetTransactionId().AsString()); >> >>>>>> m_TransportManager->EnqueueEvent( new SIPTransportEvent( msg, >>>>>> >>>> eventType ) );*/ >>>> >>>>>> } >>>>>> >>>>>> there is not leak. But if I modify the code like this: >>>>>> >>>>>> >>>>>> void SIPStack::EnqueueTransportWriteEvent( >>>>>> >>>>>> const SIPMessage & message, >>>>>> SIPTransportEvent::Type eventType >>>>>> ) >>>>>> { >>>>>> if( m_IsTerminating ) >>>>>> return; >>>>>> >>>>>> SIPMessage * msg = new SIPMessage(message); >>>>>> >>>>>> /*msg->SetInternalHeader( "TRN-ID", >>>>>> >>>> msg->GetTransactionId().AsString()); >>>> >>>>>> m_TransportManager->EnqueueEvent( new SIPTransportEvent( msg, >>>>>> >>>> eventType ) );*/ >>>> >>>>>> delete msg; >>>>>> } >>>>>> >>>>>> the leak shows up. I don't understand why this happens if I'm >>>>>> >>>> deleting the msg. >>>> >>>>>> What i said in the last mail about CSeq is not seem to be the >>>>>> >> reason. >> >>>>>> Any idea? Thanks for your help. >>>>>> Gustavo >>>>>> >>>>>> >>>>>> From: cur...@ho...Subject: RE: [OpenSIPStack] >>>>>> >>>> Memory Leak in Proxy and Full ModeDate: Fri, 30 May 2008 17:04:37 >>>> >> +0200 >> >>>>>> Hi Joegen: I found that the memory leak is when you create a >>>>>> >>>> SIPMessage from other SIPMessage or from a OString and then you >>>> >> delete >> >>>> de message. In the case of the 200 Ok the message is created in: >>>> >>>> >> ProxySessionManager::OnOrphanedMessage()SIPUserAgent::TransportWrite()SIPStack::EnqueueTransportWriteEvent(){SIPMessage >> >> >>>> * msg = new SIPMessage(message); The memory leak is because the CSeq >>>> header. If i comment : SIPMessage::SIPMessage(const SIPMessage & >>>> msg)SIPMessage & SIPMessage::operator=(const SIPMessage & msg)void >>>> SIPMessage::AssignContents(SIPMessage & msg){ /*if( m_CSeq != NULL ) >>>> msg.m_CSeq = static_cast<CSeq*>(m_CSeq->Clone());*/} There is no more >>>> memory leak. I try to replace: SIPMessage * msg = new >>>> SIPMessage(message); ----> SIPMessage * msg = new >>>> SIPMessage(message.AsString()); but the memory leak still exist >>>> >> unless >> >>>> i comment: void SIPMessage::Finalize(){ /*if( m_CSeq == NULL ) { >>>> m_CSeq = new CSeq( h ); }*/} That's what i found so far. Gustavo >>>> >>>>>> >>>>>>> Date: Fri, 30 May 2008 11:26:11 +0800> To: >>>>>>> >>>> cur...@gm...; ope...@li...> >>>> Subject: Re: [OpenSIPStack] Memory Leak in Proxy and Full Mode> From: >>>> joe...@gm...> > Hi Gustavo,> > Take a look at void >>>> ProxySession::OnFinalResponse( SIPMessage & message > ) method. Can >>>> you verify if the object created in:> > manager.CreateTuple( >>>> m_OriginalInvite, m_RoutedInvite, 10 );> > Actually expires after 10 >>>> seconds?> > Putting a breakpoint at >>>> ProxySessionTupleManager::Tuple::~Tuple() should > be enough to >>>> confirm it.> > Also by any chance, did you accidentally think that >>>> this tuple is the > mem leak because it was created after the >>>> transaction?> > Joegen> > > Gustavo Curetti wrote:> > Hi Joegen> > >>>> >>>> I found a memory leak when the OpenSBC is configured in Proxy or Full >>>> mode. When i send a 200 Ok for example in B2B mode there is no memory >>>> leak, but when i send a 200 ok in Proxy or Full mode, some memory is >>>> taken and never released. I attach the logs. The OpenSBC >>>> (192.168.0.202:5070) is running under Windows.> > > > Thanks for your >>>> help> > > > Gustavo> > >>>> _________________________________________________________________> > >>>> Ingresá ya a MSN Deportes y enterate de las últimas novedades del >>>> mundo deportivo.> > http://msn.foxsports.com/fslasc/> > >>>> >>>> >> ------------------------------------------------------------------------> >> >> -------------------------------------------------------------------------> >> >> >>>>> This SF.net email is sponsored by: Microsoft> > Defy all >>>>> >> challenges. >> >>>> Microsoft(R) Visual Studio 2008.> > >>>> http://clk.atdmt.com/MRT/go/vse0120000070mrt/direct/01/> > >>>> >>>> >> ------------------------------------------------------------------------> >> >>>>>>> _______________________________________________> > >>>>>>> >>>> opensipstack-devel mailing list> > >>>> ope...@li...> > >>>> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel> > >>>> >>>> >>>> >> ------------------------------------------------------------------------> >> >>>>>>> No virus found in this incoming message.> > Checked by AVG. > > >>>>>>> >>>> Version: 7.5.524 / Virus Database: 269.24.1/1463 - Release Date: >>>> 5/23/2008 3:36 PM> > > > >>>> >>>>>> Ingresá ya a MSN en Concierto y disfrutá los recitales en vivo de >>>>>> >>>> tus artistas favoritos. MSN en Concierto >>>> >>>>>> _________________________________________________________________ >>>>>> Descargá ya gratis y viví la experiencia Windows Live. >>>>>> http://www.descubrewindowslive.com/latam/index.html >>>>>> >>>>>> >> ------------------------------------------------------------------------- >> >>>>>> This SF.net email is sponsored by: Microsoft >>>>>> Defy all challenges. Microsoft(R) Visual Studio 2008. >>>>>> http://clk.atdmt.com/MRT/go/vse0120000070mrt/direct/01/ >>>>>> _______________________________________________ >>>>>> opensipstack-devel mailing list >>>>>> ope...@li... >>>>>> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>> >>>>> >>>>> >> ------------------------------------------------------------------------- >> >>>>> This SF.net email is sponsored by: Microsoft >>>>> Defy all challenges. Microsoft(R) Visual Studio 2008. >>>>> http://clk.atdmt.com/MRT/go/vse0120000070mrt/direct/01/ >>>>> _______________________________________________ >>>>> opensipstack-devel mailing list >>>>> ope...@li... >>>>> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel >>>>> >>>> >>>> >> ------------------------------------------------------------------------ >> >>>> Descargá ya gratis y viví la experiencia Windows Live. Descubre >>>> Windows Live <http://www.descubrewindowslive.com/latam/index.html> >>>> >>>> >> ------------------------------------------------------------------------ >> >>>> Internal Virus Database is out-of-date. >>>> Checked by AVG. >>>> Version: 7.5.524 / Virus Database: 269.24.1/1463 - Release Date: >>>> >> 5/23/2008 3:36 PM >> >>> >> ------------------------------------------------------------------------ >> Ingresá ya a MSN Deportes y enterate de las últimas novedades del >> mundo deportivo. MSN Deportes <http://msn.foxsports.com/fslasc/> >> ------------------------------------------------------------------------ >> >> Internal Virus Database is out-of-date. >> Checked by AVG. >> Version: 7.5.524 / Virus Database: 269.24.1/1463 - Release Date: 5/23/2008 3:36 PM >> >> > > > > ------------------------------------------------------------------------- > Check out the new SourceForge.net Marketplace. > It's the best place to buy or sell services for > just about anything Open Source. > http://sourceforge.net/services/buy/index.php > _______________________________________________ > opensipstack-devel mailing list > ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel > > > |
|
From: Joegen E. B. <joe...@gm...> - 2008-06-19 00:50:12
|
Right! I've found m_MinSE is not getting deleted in
SIPMessage::CleanUp(). Patched this in CVS.
Joegen
Gustavo Curetti wrote:
> Hi Joegen
>
> The modification doesn't solve the memory issues. I continue searching
> for the memory leak.
>
> A new case is attached and this one appear too when debugging with
> Microsot Visual.
>
> Originally, i sent the attached Invite every 250 ms and I set the
> timer B and H in 20 ms:
>
> #define SIP_TIMER_B 20
>
> #define SIP_TIMER_H 20
>
> Then I change the code
> of B2BUserAgent::Registrar::ProcessUpperRegKeepAlive()
> for making easier to replicate the leak:
>
>
> void B2BUserAgent::Registrar::ProcessUpperRegKeepAlive()
> {
> while( !m_UpperRegSync.Wait( 250 ) )
> {
> OString testRequest =
> "INVITE sip:5435155555@192.168.0.5:5060 SIP/2.0\r\nContact:
> <sip:4284623@192.168.0.10:5060>\r\nCSeq: 101 INVITE\r\nFrom:
> <sip:4284623@192.168.0.10>;tag=5A3745C-2418\r\nTo:
> <sip:55555555@192.168.0.206>\r\nVia: SIP/2.0/UDP
> 192.168.0.206:5060;branch=z9hG4bK63028de3a6b7743a\r\nVia: SIP/2.0/UDP
> 192.168.0.10:5060\r\nRecord-Route:
> <sip:192.168.0.206:5060;lr>\r\nAllow: INVITE, OPTIONS, BYE, CANCEL,
> ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO\r\nUser-Agent:
> Cisco-SIPGateway/IOS-12.x\r\nCall-Id:
> 3738EB25-278011DD-B92F90A6-C0EF6BE3@192.168.0.10\r\nMax-Forwards
> <mailto:3738EB25-278011DD-B92F90A6-C0EF6BE3@192.168.0.10%5Cr%5CnMax-Forwards>:
> 6\r\nExpires: 180\r\nContent-Length: 235\r\ndate: Thu, 22 May 2008
> 21:52:32 GMT\r\nsupported: timer\r\nmin-se: 1800\r\ncisco-guid:
> 926237238-662704605-3106705574-3236916195\r\nremote-party-id:
> <sip:4284623@192.168.0.10>;party=calling;screen=no;privacy=off\r\ntimestamp:
> 1211493152\r\nallow-events: telephone-event\r\ncontent-type:
> application/sdp\r\n\r\nv=0\r\no=CiscoSystemsSIP-GW-UserAgent 7402 717
> IN IP4 192.168.0.10\r\ns=SIP Call\r\nc=IN IP4 192.168.0.10\r\nt=0
> 0\r\nm=audio 19298 RTP/AVP 0 19\r\nc=IN IP4 192.168.0.10\r\na=rtpmap:0
> PCMU/8000\r\na=rtpmap:19 CN/8000\r\na=ptime:20";
> testRequest = ParserTools::LineFeedSanityCheck( testRequest );
> SIPMessage * msg = new SIPMessage( testRequest );
>
> OString addrStr = "192.168.0.147";
> OString portStr = "10000";
> SIPHeader rcvAddr( "RCVADDR", addrStr );
> SIPHeader rcvPort( "RCVPORT", portStr );
> SIPHeader rcvTran( "RCVTRAN", "udp" );
>
> msg->AddInternalHeader( rcvAddr );
> msg->AddInternalHeader( rcvPort );
> msg->AddInternalHeader( rcvTran );
> msg->SetInterfaceAddress( "192.168.0.202" );
> msg->SetInterfacePort( 5070 );
>
> OStringStream traceStream;
>
> traceStream << "<<< "
> << msg->GetStartLine() << " "
> << " SRC: " << addrStr << ":" << portStr << ":UDP"
> << " enc=" << msg->IsEncrypted()
> << " bytes=1103";
>
> OStringStream strPacket;
> strPacket << *msg;
> COMPOUND_LOG_CONTEXT( LogInfo(), msg->GetCallId(),
> traceStream.str(), LogDebugHigh(), strPacket );
>
> SIPTransport::NotifyRead( traceStream.str() );
>
> if( msg->IsInvite() )
> {
> SIPMessage * trying = new SIPMessage();
> msg->CreateResponse( *trying, SIPMessage::Code100_Trying );
> Via via;
> msg->GetViaAt(0, via );
> if( via.IsBehindNAT() )
> {
> SIPURI srcURI;
> srcURI.SetHost(addrStr);
> srcURI.SetPort(portStr);
> trying->SetSendAddress(srcURI);
> }
> if( msg->IsEncrypted() )
> trying->SetEncryption( TRUE );
> GetTransportManager()->ProcessOutbound( trying );
> }
>
> GetTransportManager()->OnTransportEvent(
> new SIPTransportEvent(
> msg,
> SIPTransportEvent::UDPPacketArrival
> ) );
>
> /*///process the keep alives here
> for( PINDEX i = 0; i < GetRegistrationDB().GetSize(); i++ )
> {
> SIPMessage reg;
> if( GetRegistrationDB().GetRegistration( i, reg ) )
> {
> if( reg.HasInternalHeader( "upper-reg" ) )
> {
> /// this is an upper reg, send a keep-alive
> /// Check the last via if its from a private IP
> Via via;
> if( reg.GetViaAt( reg.GetViaSize() - 1, via ) )
> {
> if( via.IsBehindNAT() )
> {
> SIPURI target;
> target.SetHost( via.GetReceiveAddress().AsSTLString() );
> target.SetPort( via.GetRPort() );
> SIPMessage keepAlive;
> RequestLine requestLine;
> requestLine.SetMethod( "KEEP-ALIVE" );
> requestLine.SetRequestURI( target );
> keepAlive.SetStartLine( requestLine );
> GetUserAgent().TransportWrite( keepAlive );
> }
> }
> }
> }
> }*/
> }
> }
>
>
> The OpenSBC is in "Proxy Only Mode" and the configuration is in
> "OpenSBC.reg" (attached).
>
> I compile the OpenSBC in Microsoft Visual C++ 2005 obtaining the exe
> attached.
>
> Any idea?
>
> Thanks for your help.
>
> Gustavo
>
>
>
> ------------------------------------------------------------------------
>
> > Date: Thu, 5 Jun 2008 14:18:35 +0800
> > To: cur...@gm...
> > Subject: Re: [OpenSIPStack] FW: Memory Leak in Proxy and Full Mode
> > From: joe...@gm...
> >
> > Hi Gustavo,
> >
> > Yes, i'm able to replicate it. For some reason, the code I #ifdefed in
> > AppendVia() below is causing it. Let me know if it solves your issues.
> > I've tried looking at what its doing but nothing is evident as to
> why it
> > would leak. If you find something, let me know
> >
> > Joegen
> >
> > BOOL SIPMessage::AppendVia(
> > const Via & header
> > )
> > {
> > GlobalLock();
> >
> > ParseViaList();
> >
> > if( m_ViaList == NULL )
> > {
> > m_ViaList = new Via::Collection();
> > m_ViaList->Append( new Via( header ) );
> > }else
> > {
> >
> > #if 0 // For some reason, this sanity check is leaking mem
> > /// sanity check
> > if( m_ViaList->GetSize() > 0 )
> > {
> > Via & topVia = (*m_ViaList)[0];
> >
> > SIPURI topViaURI = topVia.GetURI();
> >
> > SIPURI newURI = header.GetURI();
> >
> > if( SIPTransport::IsTheSameAddress( topViaURI, newURI, TRUE ) )
> > return FALSE;
> > }
> > #endif
> >
> > ///we append it on top
> > Via::Collection oldViaList = *m_ViaList;
> > oldViaList.MakeUnique();
> > delete m_ViaList;
> >
> > m_ViaList = new Via::Collection();
> > m_ViaList->Append( new Via( header ) );
> >
> > for( PINDEX i = 0; i < oldViaList.GetSize(); i++ )
> > m_ViaList->Append( new Via( oldViaList[i] ) );
> >
> > m_ViaList->MakeUnique();
> > }
> >
> >
> > return TRUE;
> > }
> >
> >
> > Gustavo Curetti wrote:
> > >
> > > Hi Joegen
> > >
> > > The destructor is called. The problem seem to be the headers like
> Via,
> > > RecordRoute, Contact, Allow, Supported (List headers).
> > >
> > > The leak is very easy to reproduce. I change the code of
> > > B2BUserAgent::Registrar::ProcessUpperRegKeepAlive:
> > >
> > >
> > > void B2BUserAgent::Registrar::ProcessUpperRegKeepAlive()
> > > {
> > > while( !m_UpperRegSync.Wait( 10 ) )
> > > {
> > >
> > > SIPMessage * msg = new SIPMessage();
> > >
> > > msg->AppendVia(Via("SIP/2.0/UDP 192.168.0.10:5060"));
> > > msg->AppendVia(Via("SIP/2.0/UDP
> > >
> 192.168.0.206:5060;branch=z9hG4bK440fdc3e04de9d10;rport=5060;received=192.168.0.206"));
> > > msg->AppendRecordRoute(RouteURI("<sip:192.168.0.206:5060;lr>"));
> > > msg->AppendContact(ContactURI("<sip:5435155555@192.168.0.5>"));
> > > msg->AppendAllow(Allow("INVITE, ACK, CANCEL, OPTIONS, BYE, REFER,
> > > SUBSCRIBE, NOTIFY"));
> > > msg->AppendSupported(Supported("replaces"));
> > >
> > > delete msg;
> > > }
> > > }
> > >
> > >
> > > I compile the OpenSBC in Microsoft Visual C++ 2005 obtaining the exe
> > > attached.
> > > I execute "OpenSBC Debug" in Windows 2003 or Windows 2000 and the
> leak
> > > is there.
> > >
> > > I put traces and SIPMessage::~SIPMessage(), SIPMessage::Cleanup() are
> > > called.
> > >
> > > Any idea?
> > >
> > > Thanks for your help.
> > >
> > > Gustavo
> > >
> > >
> ------------------------------------------------------------------------
> > >
> > > > Date: Mon, 2 Jun 2008 10:19:08 +0800
> > > > To: cur...@gm...;
> ope...@li...
> > > > From: joe...@gm...
> > > > Subject: Re: [OpenSIPStack] FW: Memory Leak in Proxy and Full Mode
> > > >
> > > > Try putting a trace before and after the "delete" statement to
> be sure
> > > > that its getting called. From what i see, assuming that SIPMessage
> > > > destructor performs proper cleanup, that a leak here can only be
> caused
> > > > by an exception occurring in the SIPMessage constructor;
> > > >
> > > > Gustavo Curetti wrote:
> > > > > Hi Joegen:
> > > > >
> > > > > The memory leak is hard to find. I send the 200 OK repeatedly but
> > > the memory leak doesn't appear if i'm debugging with Microsot Visual
> > > 2005, but the leak appears if i'm running the OpenSBC like a service.
> > > > >
> > > > > If I comment:
> > > > >
> > > > >
> > > > > void SIPStack::EnqueueTransportWriteEvent(
> > > > > const SIPMessage & message,
> > > > > SIPTransportEvent::Type eventType
> > > > > )
> > > > > {
> > > > > if( m_IsTerminating )
> > > > > return;
> > > > >
> > > > > /*SIPMessage * msg = new SIPMessage(message);
> > > > >
> > > > > msg->SetInternalHeader( "TRN-ID",
> msg->GetTransactionId().AsString());
> > > > > m_TransportManager->EnqueueEvent( new SIPTransportEvent( msg,
> > > eventType ) );*/
> > > > > }
> > > > >
> > > > > there is not leak. But if I modify the code like this:
> > > > >
> > > > >
> > > > > void SIPStack::EnqueueTransportWriteEvent(
> >
> > > > > const SIPMessage & message,
> > > > > SIPTransportEvent::Type eventType
> > > > > )
> > > > > {
> > > > > if( m_IsTerminating )
> > > > > return;
> > > > >
> > > > > SIPMessage * msg = new SIPMessage(message);
> > > > >
> > > > > /*msg->SetInternalHeader( "TRN-ID",
> > > msg->GetTransactionId().AsString());
> > > > > m_TransportManager->EnqueueEvent( new SIPTransportEvent( msg,
> > > eventType ) );*/
> > > > > delete msg;
> > > > > }
> > > > >
> > > > > the leak shows up. I don't understand why this happens if I'm
> > > deleting the msg.
> > > > > What i said in the last mail about CSeq is not seem to be the
> reason.
> > > > > Any idea? Thanks for your help.
> > > > > Gustavo
> > > > >
> > > > >
> > > > > From: cur...@ho...Subject: RE: [OpenSIPStack]
> > > Memory Leak in Proxy and Full ModeDate: Fri, 30 May 2008 17:04:37
> +0200
> > > > >
> > > > >
> > > > > Hi Joegen: I found that the memory leak is when you create a
> > > SIPMessage from other SIPMessage or from a OString and then you
> delete
> > > de message. In the case of the 200 Ok the message is created in:
> > >
> ProxySessionManager::OnOrphanedMessage()SIPUserAgent::TransportWrite()SIPStack::EnqueueTransportWriteEvent(){SIPMessage
>
> > > * msg = new SIPMessage(message); The memory leak is because the CSeq
> > > header. If i comment : SIPMessage::SIPMessage(const SIPMessage &
> > > msg)SIPMessage & SIPMessage::operator=(const SIPMessage & msg)void
> > > SIPMessage::AssignContents(SIPMessage & msg){ /*if( m_CSeq != NULL )
> > > msg.m_CSeq = static_cast<CSeq*>(m_CSeq->Clone());*/} There is no more
> > > memory leak. I try to replace: SIPMessage * msg = new
> > > SIPMessage(message); ----> SIPMessage * msg = new
> > > SIPMessage(message.AsString()); but the memory leak still exist
> unless
> > > i comment: void SIPMessage::Finalize(){ /*if( m_CSeq == NULL ) {
> > > m_CSeq = new CSeq( h ); }*/} That's what i found so far. Gustavo
> > > > >
> > > > >
> > > > >> Date: Fri, 30 May 2008 11:26:11 +0800> To:
> > > cur...@gm...; ope...@li...>
> > > Subject: Re: [OpenSIPStack] Memory Leak in Proxy and Full Mode> From:
> > > joe...@gm...> > Hi Gustavo,> > Take a look at void
> > > ProxySession::OnFinalResponse( SIPMessage & message > ) method. Can
> > > you verify if the object created in:> > manager.CreateTuple(
> > > m_OriginalInvite, m_RoutedInvite, 10 );> > Actually expires after 10
> > > seconds?> > Putting a breakpoint at
> > > ProxySessionTupleManager::Tuple::~Tuple() should > be enough to
> > > confirm it.> > Also by any chance, did you accidentally think that
> > > this tuple is the > mem leak because it was created after the
> > > transaction?> > Joegen> > > Gustavo Curetti wrote:> > Hi Joegen> >
> > >
> > > I found a memory leak when the OpenSBC is configured in Proxy or Full
> > > mode. When i send a 200 Ok for example in B2B mode there is no memory
> > > leak, but when i send a 200 ok in Proxy or Full mode, some memory is
> > > taken and never released. I attach the logs. The OpenSBC
> > > (192.168.0.202:5070) is running under Windows.> > > > Thanks for your
> > > help> > > > Gustavo> >
> > > _________________________________________________________________> >
> > > Ingresá ya a MSN Deportes y enterate de las últimas novedades del
> > > mundo deportivo.> > http://msn.foxsports.com/fslasc/> >
> > >
> ------------------------------------------------------------------------>
> > > >> >
> > >
> ------------------------------------------------------------------------->
>
> > > > This SF.net email is sponsored by: Microsoft> > Defy all
> challenges.
> > > Microsoft(R) Visual Studio 2008.> >
> > > http://clk.atdmt.com/MRT/go/vse0120000070mrt/direct/01/> >
> > >
> ------------------------------------------------------------------------>
> > > >> > _______________________________________________> >
> > > opensipstack-devel mailing list> >
> > > ope...@li...> >
> > > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel> >
> > >
> > >
> ------------------------------------------------------------------------>
> > > >> > No virus found in this incoming message.> > Checked by AVG. > >
> > > Version: 7.5.524 / Virus Database: 269.24.1/1463 - Release Date:
> > > 5/23/2008 3:36 PM> > > >
> > > > >>
> > > > >
> > > > > Ingresá ya a MSN en Concierto y disfrutá los recitales en vivo de
> > > tus artistas favoritos. MSN en Concierto
> > > > > _________________________________________________________________
> > > > > Descargá ya gratis y viví la experiencia Windows Live.
> > > > > http://www.descubrewindowslive.com/latam/index.html
> > > > >
> > >
> -------------------------------------------------------------------------
> > > > > This SF.net email is sponsored by: Microsoft
> > > > > Defy all challenges. Microsoft(R) Visual Studio 2008.
> > > > > http://clk.atdmt.com/MRT/go/vse0120000070mrt/direct/01/
> > > > > _______________________________________________
> > > > > opensipstack-devel mailing list
> > > > > ope...@li...
> > > > > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel
> > > > >
> > > > >
> > > > >
> > > > >
> > > >
> > > >
> > > >
> > > >
> > >
> -------------------------------------------------------------------------
> > > > This SF.net email is sponsored by: Microsoft
> > > > Defy all challenges. Microsoft(R) Visual Studio 2008.
> > > > http://clk.atdmt.com/MRT/go/vse0120000070mrt/direct/01/
> > > > _______________________________________________
> > > > opensipstack-devel mailing list
> > > > ope...@li...
> > > > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel
> > >
> > >
> > >
> ------------------------------------------------------------------------
> > > Descargá ya gratis y viví la experiencia Windows Live. Descubre
> > > Windows Live <http://www.descubrewindowslive.com/latam/index.html>
> > >
> ------------------------------------------------------------------------
> > >
> > > Internal Virus Database is out-of-date.
> > > Checked by AVG.
> > > Version: 7.5.524 / Virus Database: 269.24.1/1463 - Release Date:
> 5/23/2008 3:36 PM
> > >
> >
> >
>
>
> ------------------------------------------------------------------------
> Ingresá ya a MSN Deportes y enterate de las últimas novedades del
> mundo deportivo. MSN Deportes <http://msn.foxsports.com/fslasc/>
> ------------------------------------------------------------------------
>
> Internal Virus Database is out-of-date.
> Checked by AVG.
> Version: 7.5.524 / Virus Database: 269.24.1/1463 - Release Date: 5/23/2008 3:36 PM
>
|
|
From: OpenSIPStack F. <ope...@op...> - 2008-06-18 23:47:17
|
Hello! I am having issues registering a trunk with my voip provider. I am using ngrep and cannot see any attempts to register at all. I have trunk ports enabled in the general config. I have this in my sip trunk configuration: |
|
From: H.Kropf <mai...@gl...> - 2008-06-18 13:07:43
|
Hi, Joegen Thanks, in the CVS 2008-06-17 this problem is resolved :) jo...@op... wrote: > Whit and H.Kropf, > > A leak in OpalFlexiTranscoder proxy class maybe causing this behavior. > OpalFlexiTranscoder was first introduced by Ilian Pinzon so that > re-invites with a media channel change my allow the codec to be changed > in mid session. OpalFlexiTranscoder, however, does not have a > destructor override. This allowed the OpalTranscoder object it created > in the heap to be forever lost. I have committed a patch for this in > CVS and is ready for testing. Let me know if if does/doesn't fix this > issue. Thanks. > > Joegen > > Joegen E. Baclor wrote: > >> Ok this bug seems legitimate. I have created a ticket for this in >> assembla. >> >> http://www.assembla.com/spaces/opensbc/tickets/21 . >> >> If there some more info you could send that might help crush this bug, >> please send them in. As a backgrounder, the Voice Age G.729 codec >> educational license only allows for one channel to be opened. This is >> the reason why OPAL statically flag its usage. Thus, it is very >> possible that this bug only became evident with G.729 but may also be >> true for other codecs as well. The only diff is other codec might be >> more forgiving than G.729. More information that would >> confirm/disprove this case, please send them in. >> >> Thanks >> >> Joegen >> >> Whit Thiele wrote: >> >> >>> I have also been able to replicate this issue. So far I've only seen it >>> happen using G.729. The first call seems to go through, but every subsequent >>> call attempt fails. I am going to try recompiling an older snapshot of the >>> CVS source I have. >>> >>> H.Kropf - Have you found out anything new on this issue? >>> >>> >>> Whit >>> >>> -----Original Message----- >>> From: ope...@li... >>> [mailto:ope...@li...] On Behalf Of >>> Joegen E. Baclor >>> Sent: Thursday, May 22, 2008 9:14 PM >>> To: H.Kropf >>> Cc: ope...@li... >>> Subject: Re: [OpenSIPStack] [SF] RTP fault >>> >>> Hi, >>> >>> Thanks for the logs. They were very helpful. It is evident that the >>> previous call did not destroy the codec properly. I will try to >>> replicate this on my system. Ilian is currently on leave so it might >>> take a while for him to catch this thread. I have created a ticket for >>> this: http://www.assembla.com/spaces/opensbc/tickets/20 >>> >>> Joegen >>> >>> H.Kropf wrote: >>> >>> >>> >>>> Hi >>>> >>>> Log (for pwlib - level 6) - in attachments >>>> >>>> >>>> Joegen E. Baclor wrote: >>>> >>>> >>>> >>>>> Hmmn. Strange. Seems like the codec from the previous call has not >>>>> been destroyed. Can you send me a maximum level log of two >>>>> consecutive calls right after fresh startup of your SoftPhone? >>>>> >>>>> H.Kropf wrote: >>>>> >>>>> >>>>> >>>>> >>>>>> Hi >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>>>> can you put a break-point in OpalTranscoder::ConvertFrames() and >>>>>>>> figure out where exactly it fails? >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>> I can :) >>>>>> >>>>>> >>>>>> bool OpalMediaPatch::Sink::WriteFrame(RTP_DataFrame & sourceFrame) >>>>>> { >>>>>> ...... >>>>>> if (!primaryCodec->ConvertFrames(sourceFrame, intermediateFrames)) >>>>>> { >>>>>> PTRACE(1, "Patch\tMedia conversion (primary) failed"); >>>>>> return false; >>>>>> } >>>>>> ...... >>>>>> } >>>>>> >>>>>> BOOL OpalTranscoder::ConvertFrames(const RTP_DataFrame & input, >>>>>> RTP_DataFrameList & output) >>>>>> { >>>>>> ....... >>>>>> return Convert(input, output[0]); >>>>>> } >>>>>> >>>>>> BOOL OpalFlexiTranscoder::Convert( const RTP_DataFrame & input, >>>>>> RTP_DataFrame & output) >>>>>> { >>>>>> return m_Transcoder->Convert( input, output ); >>>>>> } >>>>>> >>>>>> BOOL OpalFramedTranscoder::Convert(const RTP_DataFrame & input, >>>>>> RTP_DataFrame & output) >>>>>> { >>>>>> .... >>>>>> while (inputLength > 0) >>>>>> { >>>>>> ...... >>>>>> if (!ConvertFrame(inputPtr, consumed, outputPtr, created)) >>>>>> return FALSE; >>>>>> ....... >>>>>> } >>>>>> ....... >>>>>> } >>>>>> >>>>>> BOOL OpalFramedTranscoder::ConvertFrame(const BYTE * inputPtr, >>>>>> PINDEX & /*consumed*/, BYTE * outputPtr, PINDEX & /*created*/) >>>>>> { >>>>>> return ConvertFrame(inputPtr, outputPtr); >>>>>> } >>>>>> >>>>>> >>>>>> >>>>>> BOOL Opal_PCM_G729::ConvertFrame(const BYTE * src, BYTE * dst) >>>>>> { >>>>>> if (voiceAgeEncoderInUse != this) return FALSE; // >>>>>> !!!!!!!!!!!! <<<=== this place >>>>>> ...... >>>>>> } >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>>> Hi, >>>>>>> >>>>>>> Are you a C++ developer? If so, can you put a break-point in >>>>>>> OpalTranscoder::ConvertFrames() and figure out where exactly it fails? >>>>>>> >>>>>>> Joegen >>>>>>> >>>>>>> H.Kropf wrote: >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>>> Hello >>>>>>>> >>>>>>>> After a last update of library from CVS (2008-05-20), my softphone >>>>>>>> (on OSS library) makes only one successful call after start. In >>>>>>>> next calls there is no voice. In PTRACE-log there are many such >>>>>>>> records >>>>>>>> >>>>>>>> Media Patch:1eb9410 PWL: [CID=0x0000] Patch Media conversion >>>>>>>> (primary) failed >>>>>>>> Media Patch:1eb9410 PWL: [CID=0x0000] Patch Media conversion >>>>>>>> (primary) failed >>>>>>>> Media Patch:1eb9410 PWL: [CID=0x0000] Patch Media conversion >>>>>>>> (primary) failed >>>>>>>> Media Patch:1eb9410 PWL: [CID=0x0000] Patch Media conversion >>>>>>>> (primary) failed >>>>>>>> >>>>>>>> >>>>>>>> In the previous version (CVS 2008-05-12) this problem did not exist >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>> ------------------------------------------------------------------------- >>> >>> >>> >>>>>>>> This SF.net email is sponsored by: Microsoft Defy all challenges. >>>>>>>> Microsoft(R) Visual Studio 2008. >>>>>>>> http://clk.atdmt.com/MRT/go/vse0120000070mrt/direct/01/ >>>>>>>> _______________________________________________ >>>>>>>> opensipstack-devel mailing list >>>>>>>> ope...@li... >>>>>>>> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>> ------------------------------------------------------------------------- >>> >>> >>> >>>>>>> This SF.net email is sponsored by: Microsoft Defy all challenges. >>>>>>> Microsoft(R) Visual Studio 2008. >>>>>>> http://clk.atdmt.com/MRT/go/vse0120000070mrt/direct/01/ >>>>>>> _______________________________________________ >>>>>>> opensipstack-devel mailing list >>>>>>> ope...@li... >>>>>>> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>> ------------------------------------------------------------------------- >>> >>> >>> >>>>>> This SF.net email is sponsored by: Microsoft Defy all challenges. >>>>>> Microsoft(R) Visual Studio 2008. >>>>>> http://clk.atdmt.com/MRT/go/vse0120000070mrt/direct/01/ >>>>>> _______________________________________________ >>>>>> opensipstack-devel mailing list >>>>>> ope...@li... >>>>>> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>> ------------------------------------------------------------------------- >>>>> >>>>> >>>>> >>> >>> >>> >>>>> This SF.net email is sponsored by: Microsoft Defy all challenges. >>>>> Microsoft(R) Visual Studio 2008. >>>>> http://clk.atdmt.com/MRT/go/vse0120000070mrt/direct/01/ >>>>> _______________________________________________ >>>>> opensipstack-devel mailing list >>>>> ope...@li... >>>>> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel >>>>> >>>>> >>>>> >>>>> >>>>> >>>> ------------------------------------------------------------------------ >>>> >>>> No virus found in this incoming message. >>>> Checked by AVG. >>>> Version: 7.5.524 / Virus Database: 269.23.21/1458 - Release Date: >>>> >>>> >>>> >>> 5/21/2008 7:21 AM >>> >>> >>> ------------------------------------------------------------------------- >>> This SF.net email is sponsored by: Microsoft >>> Defy all challenges. Microsoft(R) Visual Studio 2008. >>> http://clk.atdmt.com/MRT/go/vse0120000070mrt/direct/01/ >>> _______________________________________________ >>> opensipstack-devel mailing list >>> ope...@li... >>> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel >>> >>> No virus found in this incoming message. >>> Checked by AVG. >>> Version: 8.0.100 / Virus Database: 269.23.21/1458 - Release Date: 5/21/2008 >>> 7:21 AM >>> >>> >>> >>> >>> >>> >> ------------------------------------------------------------------------- >> Check out the new SourceForge.net Marketplace. >> It's the best place to buy or sell services for >> just about anything Open Source. >> http://sourceforge.net/services/buy/index.php >> _______________________________________________ >> opensipstack-devel mailing list >> ope...@li... >> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel >> >> >> >> >> > > > ------------------------------------------------------------------------- > Check out the new SourceForge.net Marketplace. > It's the best place to buy or sell services for > just about anything Open Source. > http://sourceforge.net/services/buy/index.php > _______________________________________________ > opensipstack-devel mailing list > ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel > > |
|
From: H.Kropf <mai...@gl...> - 2008-06-18 13:03:46
|
Hi, It is necessary to correct file "VoiceFileChannel.h" . #ifdef WIN32 // for win32 builds, avoid function name wrong redefinition (PlaySoundA or PlaySoundW instead of PlaySound) #undef PlaySound #endif instead of #ifdef WIN32 #ifdef UNICODE // avoid function name conflicts for win32 unicode builds #undef PlaySound #endif #endif Function ms:: PlaySound gets to library OSS either as PlaySoundA or as PlaySoundW, having accepted "#define Playsound" from file VisualStudio/MMSystem.h. But in it there is no necessity because function ms:: PlaySound has no parametres of type LPCTSTR and does not require different names for ANSI and UNICODE. That the program using library OSS, was not mistaken at different "character set" IMHO is necessary this patch. Joegen E. Baclor wrote: > I haven't tested it but your patch looks ok to me. This is already in > CVS. Thanks. > > * $Log: VoiceFileChannel.cxx,v $ > * Revision 1.10 2008/06/04 00:03:18 joegenbaclor > * Fixed PlaySound name conflict bug for UNICODE definition - Thanks to > helgikropp > > H.Kropf wrote: > >> Hello >> >> Please substitute the corrected files (in attachments) >> VoiceFileChannel.h and VoiceFileChannel.cxx in CVS. >> >> This patch resolved PlaySound name conflict for compilation >> with UNICODE definition. >> ------------------------------------------------------------------------ >> >> ------------------------------------------------------------------------- >> This SF.net email is sponsored by: Microsoft >> Defy all challenges. Microsoft(R) Visual Studio 2008. >> http://clk.atdmt.com/MRT/go/vse0120000070mrt/direct/01/ >> ------------------------------------------------------------------------ >> >> _______________________________________________ >> opensipstack-devel mailing list >> ope...@li... >> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel >> ------------------------------------------------------------------------ >> >> Internal Virus Database is out-of-date. >> Checked by AVG. >> Version: 7.5.524 / Virus Database: 269.24.1/1463 - Release Date: 5/23/2008 3:36 PM >> > > > ------------------------------------------------------------------------- > Check out the new SourceForge.net Marketplace. > It's the best place to buy or sell services for > just about anything Open Source. > http://sourceforge.net/services/buy/index.php > _______________________________________________ > opensipstack-devel mailing list > ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel > > |
|
From: OpenSIPStack F. <ope...@op...> - 2008-06-18 02:49:39
|
Hi,
Thanks for confirming. Since I can't replicate this locally, it seems this problem is specific to your box. It will be of great value if we can pinpoint where exactly it fails in SIPTransport::Resolve() so we can give the library a defense mechanism to prevent this behavior. To give you a clue where to look next. A lot of the stuffs in resolving uris happen in
[http://SIPSrvRecord.cxx|http://SIPSrvRecord.cxx]
BOOL SIPSrvRecord::Resolve(
const OString & _domain,
const OString & _proto,
WORD port
);
So if it SIPTransport::Resolve fails somewhere, this is likely the place where it will happen.
Joegen
Joegen
> {quote:title=cam100 wrote:}{quote}
> Hello Joegen,
>
> here is what I get from the logs now.
>
>
> 337143:14:56.132 INF: [CID=0x0d58] <<< REGISTER sip:sip.fakedomain.net SIP/2.0 SRC: 192.168.10.3:10164:UDP enc=0 bytes=576
> 337143:14:56.143 INF: [CID=0x0d58] *** CREATED *** PROXY Session ZDY1MWMxYjg4NWM5Nzc1M2ZmMGY2MjM1NDI5ZmQ1N2E.
> 337143:14:56.156 INF: [CID=0x0d58] >>> SIP/2.0 100 Trying DST: 192.168.10.3:10164:UDP SRC: 85.xxx.x.:5060 enc=0 bytes=388
> 337143:14:56.161 INF: [CID=0x0d58] Routing REGISTER for URI sip:862...@si...
> 337143:14:56.163 PWL: [CID=0x0000] -->> From: sip:862...@si... Target: REGISTER sip:862...@so...;domain=softswitch.mydomain.net
> 337143:14:56.166 INF: [CID=0x0d58] *** UPPER REGISTRATION ENABLE *** for sip:862...@si...
> 337143:14:56.184 INF: [CID=0x0d58] >>> REGISTER sip:softswitch.mydomain.net SIP/2.0 DST: 85.xxx.x.85:5060:UDP SRC: 85.xxx.x.:5060 enc=0 bytes=785
> 337143:14:56.198 INF: [CID=0x0d58] <<< SIP/2.0 401 Unauthorized SRC: 85.xxx.x.85:5060:UDP enc=0 bytes=697
> --> those are the new debugging messages:
> 337143:14:56.267 PWL: [CID=0x0000] return false from SIPTransport::Resolve
> 337143:14:56.267 PWL: [CID=0x0000] return FALSE from SIPTransportManager::IsLocalAddressAndPort when calling SIPTransport::Resolve whith uri: sip: 85.xxx.x.:5060;iid=13477;branch=z9hG4bK7649f5cfed3add11880c857f5cf1f22e-619874e34902600b7e5781e691407b49;uas-addr=85.xxx.x.85;rport=5060;received=85.xxx.x. addr: 127.0.0.1 port: 0
> 337143:14:56.267 PWL: [CID=0x0000] FATAL !!! Unable to POP Top-most Via sip: 85.xxx.x.:5060;iid=13477;branch=z9hG4bK7649f5cfed3add11880c857f5cf1f22e-619874e34902600b7e5781e691407b49;uas-addr=85.xxx.x.85;rport=5060;received=85.xxx.x.
|
|
From: lucas m. <mar...@gm...> - 2008-06-17 19:26:37
|
Joegen, I was working and doing some more testes and the problem has disappeared when i removed the g729 codec as the default one, may be there is a bug or some miss configuration there. Let me know if you still want the whole debug. Thanks On Thu, Jun 12, 2008 at 8:18 PM, jo...@op... < joe...@gm...> wrote: > Hi Lucas, > > I do not have access to a Vista license currently. Perhaps the level 5 > PWLIB and SIP logs would tell us something. Can you send them off-list? > > Joegen > > > lucas martinez wrote: > > Joegen, > > Do you have any advise to help me working around on this matter. > Thanks > > > > On Thu, Jun 5, 2008 at 8:34 AM, lucas martinez <mar...@gm...> > wrote: > > > > > >> Thanks for answer Joegen, im just wondering if this could help to avoid > >> this kind of problem. Do you know how to solve this problem, a work > around > >> or what do i need to check? > >> > >> Thanks. > >> > >> > >> On Wed, Jun 4, 2008 at 9:17 PM, Joegen E. Baclor < > joe...@gm...> > >> wrote: > >> > >> > >>> Are you saying that the new pwlib with direct sound support has no > >>> problem in vista? > >>> > >>> > >>> lucas martinez wrote: > >>> > >>>> Hi, > >>>> I´m using the ATLSIP.DLL and when i install the application in a PC > with > >>>> Windows Vista the audio is very bad, i have been looking for some > >>>> information here and i found that seting SetSoundChannelBufferDepth( > 10 > >>>> > >>> ) > >>> > >>>> instead of 5, but we still have the same problem. > >>>> I found a new version of PWLIB(1.12.0) which support DirectSound, is > >>>> > >>> this > >>> > >>>> too hard to adapt, just wondering? > >>>> > >>>> Thank in advance. > >>>> > >>>> Lucas. > >>>> > >>>> > >>> > ------------------------------------------------------------------------- > >>> > >>>> Check out the new SourceForge.net Marketplace. > >>>> It's the best place to buy or sell services for > >>>> just about anything Open Source. > >>>> http://sourceforge.net/services/buy/index.php > >>>> _______________________________________________ > >>>> opensipstack-devel mailing list > >>>> ope...@li... > >>>> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel > >>>> > >>>> > >>>> > >>>> > >>> > >>> > ------------------------------------------------------------------------- > >>> Check out the new SourceForge.net Marketplace. > >>> It's the best place to buy or sell services for > >>> just about anything Open Source. > >>> http://sourceforge.net/services/buy/index.php > >>> _______________________________________________ > >>> opensipstack-devel mailing list > >>> ope...@li... > >>> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel > >>> > >>> > >> > > ------------------------------------------------------------------------- > > Check out the new SourceForge.net Marketplace. > > It's the best place to buy or sell services for > > just about anything Open Source. > > http://sourceforge.net/services/buy/index.php > > _______________________________________________ > > opensipstack-devel mailing list > > ope...@li... > > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel > > > > > > > > > > > > ------------------------------------------------------------------------- > Check out the new SourceForge.net Marketplace. > It's the best place to buy or sell services for > just about anything Open Source. > http://sourceforge.net/services/buy/index.php > _______________________________________________ > opensipstack-devel mailing list > ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel > |
|
From: OpenSIPStack F. <ope...@op...> - 2008-06-17 15:29:14
|
Hello Joegen, here is what I get from the logs now. 337143:14:56.132 INF: [CID=0x0d58] <<< REGISTER sip:sip.fakedomain.net SIP/2.0 SRC: 192.168.10.3:10164:UDP enc=0 bytes=576 337143:14:56.143 INF: [CID=0x0d58] *** CREATED *** PROXY Session ZDY1MWMxYjg4NWM5Nzc1M2ZmMGY2MjM1NDI5ZmQ1N2E. 337143:14:56.156 INF: [CID=0x0d58] >>> SIP/2.0 100 Trying DST: 192.168.10.3:10164:UDP SRC: 85.xxx.x.:5060 enc=0 bytes=388 337143:14:56.161 INF: [CID=0x0d58] Routing REGISTER for URI sip:862...@si... 337143:14:56.163 PWL: [CID=0x0000] -->> From: sip:862...@si... Target: REGISTER sip:862...@so...;domain=softswitch.mydomain.net 337143:14:56.166 INF: [CID=0x0d58] *** UPPER REGISTRATION ENABLE *** for sip:862...@si... 337143:14:56.184 INF: [CID=0x0d58] >>> REGISTER sip:softswitch.mydomain.net SIP/2.0 DST: 85.xxx.x.85:5060:UDP SRC: 85.xxx.x.:5060 enc=0 bytes=785 337143:14:56.198 INF: [CID=0x0d58] <<< SIP/2.0 401 Unauthorized SRC: 85.xxx.x.85:5060:UDP enc=0 bytes=697 --> those are the new debugging messages: 337143:14:56.267 PWL: [CID=0x0000] return false from SIPTransport::Resolve 337143:14:56.267 PWL: [CID=0x0000] return FALSE from SIPTransportManager::IsLocalAddressAndPort when calling SIPTransport::Resolve whith uri: sip: 85.xxx.x.:5060;iid=13477;branch=z9hG4bK7649f5cfed3add11880c857f5cf1f22e-619874e34902600b7e5781e691407b49;uas-addr=85.xxx.x.85;rport=5060;received=85.xxx.x. addr: 127.0.0.1 port: 0 337143:14:56.267 PWL: [CID=0x0000] FATAL !!! Unable to POP Top-most Via sip: 85.xxx.x.:5060;iid=13477;branch=z9hG4bK7649f5cfed3add11880c857f5cf1f22e-619874e34902600b7e5781e691407b49;uas-addr=85.xxx.x.85;rport=5060;received=85.xxx.x. |
|
From: OpenSIPStack F. <ope...@op...> - 2008-06-17 13:33:55
|
Hello Joegen,
many thanks for your reply and pointing me into the right direction. You have been perfectly right with your hunch - the problem is that the OpenSBC cannot pop the first via header. I have digged a bit deeper into the source code and added some traces to the functions where I suspect the error.
>From my point of view it goes the following way :
1. ProxySession.cxx --> GetTransportManager()->IsLocalAddressAndPort( via.GetURI() ) is called
since this call fails, response.PopTopVia() is not executed and the message is not routed back to the client
2. Within SIPTransportManager::IsLocalAddressAndPort the function SIPTransport::Resolve( uri, addr, port ) is called
- the call for this function fails and FALSE is returned
3. SIPTransport::Resolve is called. I' m still trying to find out at which point it fails in this function. I will provide an update as soon as I know.
Regards,
Andre
|
|
From: OpenSIPStack F. <ope...@op...> - 2008-06-17 02:31:39
|
It turned out it's already exposed in ATLSIP
HRESULT SetAudioJitterDelay([in] ULONG minDelay, [in] ULONG maxDelay);
just call atlsip.SetAudioJitterDelay( 10, 100 ); or something like that.
Joegen
> {quote:title=joegen wrote:}{quote}
> Hi,
>
> For the meantime you may uncomment (or put your own range in ms) the line that goes:
>
> SetAudioJitterDelay(100, 1000);
>
> in SoftPhone.cxx. Its inside the SoftPhoneManager::SoftPhoneManager() constructor block.
>
> I'll have the setter handy in ATLSIP in a shor while
>
> Joegen
>
>
> > {quote:title=Guest wrote:}{quote}
> > Hi.
> >
> > We are currently considering wether we should use OSS for a small client application (possibly using ATLSIP.dll). I've been wondering if it is possible / how difficult it is to change the buffer size because we need smaller delays (this is for LAN calls only not the internet).
> > I've been recording delays of nearly 300ms which is probably ok through the internet, but is a problem for LAN connections at call centers,.. -- I did measure this by enabling recording on our PBX and then using a SIP phone [snom 360] on one end and a PC with Solegy phone / OSSPhone on the other end. At the pc I put a cable from microphone to headset jack (with a condensator and a resistor ;)) so that we can measure the delay properly by just having a look at the wave file.
> >
> > Please let me know if this is maybe just a parameter somewhere or not configurable at all.
> >
> > Regards,
> >
> > Mit freundlichen Grüßen
> > Thomas Raschbacher
> > ____________________________________________
> > itCampus Technology GmbH
> > Österreich * Deutschland * Italien
> > Dresdner Straße 45 /DG
> > 1200 Wien
> > tho...@it...
> > Tel: +43 (1) 890 22 82 - 58
> > Fax: +43 (1) 890 22 82 - 958
> > http://www.itctec.com
> > UID: ATU 6339 0618
> > Firmenbuchnr: FN292598t, Handelsgericht Wien
> > Geschäftsführer: Andreas Günser, Andreas Lassmann
> > Joint Venture von itCampus und MEC
> >
> > itCampus Gruppe
> > Deutschland * Großbritannien * Italien * Österreich * Schweiz * Slowakei
> > http://www.itcampus.eu
> >
> >
> > -------------------------------------------------------------------------
> > Check out the new SourceForge.net Marketplace.
> > It's the best place to buy or sell services for
> > just about anything Open Source.
> > http://sourceforge.net/services/buy/index.php
> > _______________________________________________
> > opensipstack-devel mailing list
> > ope...@li...
> > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel
> >
|
|
From: OpenSIPStack F. <ope...@op...> - 2008-06-17 02:10:28
|
Hi, Try http://www.live555.com/openRTSP/ Joegen > {quote:title=Guest wrote:}{quote} > Hi all , > > I am new to blog , so please help me in this issue. > > i need build a RTSP server and client which has to start with basics > connection establishment and teardown .. process > > let me know if their is server to download and install and iam usind fedora > as my os .... > > please ............. > > -- > With regards > Prasad > ------------------------------------------------------------------------- > Check out the new SourceForge.net Marketplace. > It's the best place to buy or sell services for > just about anything Open Source. > http://sourceforge.net/services/buy/index.php > _______________________________________________ > opensipstack-devel mailing list > ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel > |
|
From: OpenSIPStack F. <ope...@op...> - 2008-06-17 02:07:59
|
Hi,
For the meantime you may uncomment (or put your own range in ms) the line that goes:
SetAudioJitterDelay(100, 1000);
in SoftPhone.cxx. Its inside the SoftPhoneManager::SoftPhoneManager() constructor block.
I'll have the setter handy in ATLSIP in a shor while
Joegen
> {quote:title=Guest wrote:}{quote}
> Hi.
>
> We are currently considering wether we should use OSS for a small client application (possibly using ATLSIP.dll). I've been wondering if it is possible / how difficult it is to change the buffer size because we need smaller delays (this is for LAN calls only not the internet).
> I've been recording delays of nearly 300ms which is probably ok through the internet, but is a problem for LAN connections at call centers,.. -- I did measure this by enabling recording on our PBX and then using a SIP phone [snom 360] on one end and a PC with Solegy phone / OSSPhone on the other end. At the pc I put a cable from microphone to headset jack (with a condensator and a resistor ;)) so that we can measure the delay properly by just having a look at the wave file.
>
> Please let me know if this is maybe just a parameter somewhere or not configurable at all.
>
> Regards,
>
> Mit freundlichen Grüßen
> Thomas Raschbacher
> ____________________________________________
> itCampus Technology GmbH
> Österreich * Deutschland * Italien
> Dresdner Straße 45 /DG
> 1200 Wien
> tho...@it...
> Tel: +43 (1) 890 22 82 - 58
> Fax: +43 (1) 890 22 82 - 958
> http://www.itctec.com
> UID: ATU 6339 0618
> Firmenbuchnr: FN292598t, Handelsgericht Wien
> Geschäftsführer: Andreas Günser, Andreas Lassmann
> Joint Venture von itCampus und MEC
>
> itCampus Gruppe
> Deutschland * Großbritannien * Italien * Österreich * Schweiz * Slowakei
> http://www.itcampus.eu
>
>
> -------------------------------------------------------------------------
> Check out the new SourceForge.net Marketplace.
> It's the best place to buy or sell services for
> just about anything Open Source.
> http://sourceforge.net/services/buy/index.php
> _______________________________________________
> opensipstack-devel mailing list
> ope...@li...
> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel
>
|
|
From: OpenSIPStack F. <ope...@op...> - 2008-06-17 01:39:07
|
oops! sorryy it should be
if( GetTransportManager()->IsLocalAddressAndPort( via.GetURI() ) )
{
response.PopTopVia();
}else
{
PTRACE( 1, "FATAL!!!! Unable to POP Top-most Via" );
}
Joegen
|
|
From: OpenSIPStack F. <ope...@op...> - 2008-06-17 01:24:38
|
Hi,
I am unable to replicate this in my box. However, I have a hunch what is causing this for you. You need to help me out understand why it is happening. Can you trace though ProxySession.cxx for me and find the following code block:
void ProxySession::OnFinalResponse( SIPMessage & message )
{
GCVERIFYREF_VOID( "ProxySession::OnFinalResponse" );
SIPMessage response = message;
Via via;
response.GetViaAt( 0, via );
if( GetTransportManager()->IsLocalAddressAndPort( via.GetURI() ) )
response.PopTopVia();
If my hunch is correct, OpenSBC is unable to POP the top-most via of the response because it did not resolve as a local address:port. Can you confirm this? Please add a PTRACE within the if block like so:
if( GetTransportManager()->IsLocalAddressAndPort( via.GetURI() ) )
{
PTRACE( 1, "FATAL!!!! Unable to POP Top-most Via" );
response.PopTopVia();
}
Let me know if it appears in your tests. If you are a developer, it would be nice if you could trace into IsLocalAddressAndPort function and tell me where it fails.
Joegen
> {quote:title=cam100 wrote:}{quote}
> Hi all,
>
>
>
>
>
> I have been searching the forum for hours now but can not find any hint what I' m doing wrong. I' m running OpenSBC in B2B upper registration mode and would like to have my clients registering whith our existing softswitch. I have set up everything according to the tutorials and howto's which can be found in this forum.
>
>
>
>
> The client sends a register message to the SBC, the SBC acknowledges that there is an upper registration route and forwards the request to the softswitch. The softswitch reguires digest authentication and sends a 401 unauthorized back. I would expect now that this message is relayed back to the original client in order to continue with the authentication.
>
>
>
> For some reason, that message never arrives at the client - therefore the client is trying and trying to authenticate.
>
>
>
>
>
>
>
> Here is what I' m seeing in the log file:
>
>
>
> 337066:25:26.471 INF: [CID=0x0e02] <<< REGISTER sip:sip.fakedomain.net SIP/2.0 SRC: 192.168.10.3:37934:UDP enc=0 bytes=576
> 337066:25:26.471 DBG: [CID=0x0e02]
> 337066:25:26.471 DBG: [CID=0x0e02] REGISTER sip:sip.fakedomain.net SIP/2.0
> 337066:25:26.471 DBG: [CID=0x0e02] From: "SoftClient01" <sip:862...@si...>;tag=6613195e
> 337066:25:26.471 DBG: [CID=0x0e02] To: "SoftClient01" <sip:862...@si...>
> 337066:25:26.471 DBG: [CID=0x0e02] Via: SIP/2.0/UDP 192.168.10.3:37934;branch=z9hG4bK-d87543-30153228e21d0017-1--d87543-
> 337066:25:26.471 DBG: [CID=0x0e02] CSeq: 1 REGISTER
> 337066:25:26.471 DBG: [CID=0x0e02] Call-ID: OWFkMTI4ODljOGQ3NjA1MDM1NzY0ZGViZDUzMjZmMmM.
> 337066:25:26.471 DBG: [CID=0x0e02] Contact: <sip:8621430679@192.168.10.3:37934;rinstance=6e999ef109459e0f>
> 337066:25:26.471 DBG: [CID=0x0e02] User-Agent: X-Lite release 1011s stamp 41150
> 337066:25:26.471 DBG: [CID=0x0e02] Expires: 3600
> 337066:25:26.471 DBG: [CID=0x0e02] Max-Forwards: 70
> 337066:25:26.471 DBG: [CID=0x0e02] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
> 337066:25:26.471 DBG: [CID=0x0e02] Content-Length: 0
> 337066:25:26.471 DBG: [CID=0x0e02]
> 337066:25:26.471 DBG: [CID=0x0e02]
> 337066:25:26.473 DBG: [CID=0x0e02] Finding transaction for REGISTER sip:sip.fakedomain.net SIP/2.0
> 337066:25:26.473 DBG: [CID=0x0e02] Setting Transaction ID to OWFkMTI4ODljOGQ3NjA1MDM1NzY0ZGViZDUzMjZmMmM.|z9hG4bK-d87543-30153228e21d0017-1--d87543-|REGISTER
> 337066:25:26.474 DBG: [CID=0x0e02]
> 337066:25:26.474 DBG: [CID=0x0e02] *** CREATING TRANSACTION (NIST) ***
> 337066:25:26.474 DBG: [CID=0x0e02] Message: REGISTER sip:sip.fakedomain.net SIP/2.0
> 337066:25:26.474 DBG: [CID=0x0e02] Call-Id: OWFkMTI4ODljOGQ3NjA1MDM1NzY0ZGViZDUzMjZmMmM.
> 337066:25:26.474 DBG: [CID=0x0e02]
> 337066:25:26.475 DTL: [CID=0x0e02] NIST(2258349004) *** CREATED *** - NIST|OWFkMTI4ODljOGQ3NjA1MDM1NzY0ZGViZDUzMjZmMmM.|z9hG4bK-d87543-30153228e21d0017-1--d87543-|REGISTER
> 337066:25:26.477 DTL: [CID=0x0e02] NIST(2258349004) Event(SIPMessage) - REGISTER sip:sip.fakedomain.net SIP/2.0
> 337066:25:26.477 DBG: [CID=0x0e02] TRANSACTION: (NIST) REGISTER sip:sip.fakedomain.net SIP/2.0 State: 0
> 337066:25:26.478 DTL: [CID=0x0e02] NIST(2258349004) StateIdle->StateTrying
> 337066:25:26.478 DBG: [CID=0x0e02] Event: SIPStack::Enqueue(REGISTER sip:sip.fakedomain.net SIP/2.0)
> 337066:25:26.480 DBG: [CID=0x0e02] Event: B2BUserAgent::ProcessEvent( REGISTER sip:sip.fakedomain.net SIP/2.0 )
> 337066:25:26.484 DBG: [CID=0x0e02] Event: Setting UA Core [Proxy] to handle event REGISTER
> 337066:25:26.484 DTL: [CID=0x0e02] Event: ---> Inbound - REGISTER sip:sip.fakedomain.net SIP/2.0
> 337066:25:26.485 DBG: [CID=0x0e02] Session CREATED
> 337066:25:26.486 INF: [CID=0x0e02] *** CREATED *** PROXY Session OWFkMTI4ODljOGQ3NjA1MDM1NzY0ZGViZDUzMjZmMmM.
> 337066:25:26.487 DBG: [CID=0x0e02] *** MESSAGE ARRIVAL *** for SIP Session OWFkMTI4ODljOGQ3NjA1MDM1NzY0ZGViZDUzMjZmMmM.
> 337066:25:26.493 DTL: [CID=0x0e02] Found NIST|OWFkMTI4ODljOGQ3NjA1MDM1NzY0ZGViZDUzMjZmMmM.|z9hG4bK-d87543-30153228e21d0017-1--d87543-|REGISTER for SIP/2.0 100 Trying
> 337066:25:26.494 DTL: [CID=0x0e02] NIST(2258349004) Event(SIPMessage) - SIP/2.0 100 Trying
> 337066:25:26.494 DBG: [CID=0x0e02] TRANSACTION: (NIST) SIP/2.0 100 Trying State: 1
> 337066:25:26.495 DTL: [CID=0x0e02] NIST(2258349004) StateTrying->StateProceeding
> 337066:25:26.499 INF: [CID=0x0e02] >>> SIP/2.0 100 Trying DST: 192.168.10.3:37934:UDP SRC: 85.xxx.x.200:5060 enc=0 bytes=388
> 337066:25:26.500 DBG: [CID=0x0e02]
> 337066:25:26.500 DBG: [CID=0x0e02] SIP/2.0 100 Trying
> 337066:25:26.500 DBG: [CID=0x0e02] From: "SoftClient01" <sip:862...@si...>;tag=6613195e
> 337066:25:26.500 DBG: [CID=0x0e02] To: "SoftClient01" <sip:862...@si...>
> 337066:25:26.500 DBG: [CID=0x0e02] Via: SIP/2.0/UDP 192.168.10.3:37934;iid=16143;branch=z9hG4bK-d87543-30153228e21d0017-1--d87543-;rport=37934;received=192.168.10.3
> 337066:25:26.500 DBG: [CID=0x0e02] CSeq: 1 REGISTER
> 337066:25:26.500 DBG: [CID=0x0e02] Call-ID: OWFkMTI4ODljOGQ3NjA1MDM1NzY0ZGViZDUzMjZmMmM.
> 337066:25:26.500 DBG: [CID=0x0e02] Server: OpenSBC v1.1.5-6
> 337066:25:26.500 DBG: [CID=0x0e02] Content-Length: 0
> 337066:25:26.500 DBG: [CID=0x0e02]
> 337066:25:26.500 DBG: [CID=0x0e02]
> 337066:25:26.504 INF: [CID=0x0e02] Routing REGISTER for URI sip:862...@si...
> 337066:25:26.505 PWL: [CID=0x0000] -->> From: sip:862...@si... Target: REGISTER sip:862...@so...;domain=softswitch.mydomain.net
> 337066:25:26.507 INF: [CID=0x0e02] *** UPPER REGISTRATION ENABLE *** for sip:862...@si...
> 337066:25:26.511 DBG: [CID=0x06cb] CREATED via=85.xxx.x.200:5060 for target=85.xxx.x.85 protocol=UDP
> 337066:25:26.513 DBG: [CID=0x0e02] Proxying request REGISTER sip:softswitch.mydomain.net SIP/2.0
> 337066:25:26.516 DBG: [CID=0x0e02] Finding transaction for REGISTER sip:softswitch.mydomain.net SIP/2.0
> 337066:25:26.516 DBG: [CID=0x0e02] Setting Transaction ID to OWFkMTI4ODljOGQ3NjA1MDM1NzY0ZGViZDUzMjZmMmM.|z9hG4bK4cba98df6938dd118b1d98e0e14f783f-69bc052885e284aa4b0d2044600ff220|REGISTER
> 337066:25:26.517 DBG: [CID=0x0e02]
> 337066:25:26.517 DBG: [CID=0x0e02] *** CREATING TRANSACTION (NICT) ***
> 337066:25:26.517 DBG: [CID=0x0e02] Message: REGISTER sip:softswitch.mydomain.net SIP/2.0
> 337066:25:26.517 DBG: [CID=0x0e02] Call-Id: OWFkMTI4ODljOGQ3NjA1MDM1NzY0ZGViZDUzMjZmMmM.
> 337066:25:26.517 DBG: [CID=0x0e02]
> 337066:25:26.518 DTL: [CID=0x0e02] NICT(2258349005) *** CREATED *** - NICT|OWFkMTI4ODljOGQ3NjA1MDM1NzY0ZGViZDUzMjZmMmM.|z9hG4bK4cba98df6938dd118b1d98e0e14f783f-69bc052885e284aa4b0d2044600ff220|REGISTER
> 337066:25:26.520 DTL: [CID=0x0e02] NICT(2258349005) Event(SIPMessage) - REGISTER sip:softswitch.mydomain.net SIP/2.0
> 337066:25:26.521 DBG: [CID=0x0e02] TRANSACTION: (NICT) REGISTER sip:softswitch.mydomain.net SIP/2.0 State: 0
> 337066:25:26.521 DTL: [CID=0x0e02] NICT(2258349005) StateIdle->StateTrying(REGISTER sip:softswitch.mydomain.net SIP/2.0)
> 337066:25:26.521 DBG: [CID=0x0e02] NICT(2258349005) Timer E( 500 ms ) STARTED
> 337066:25:26.521 DBG: [CID=0x0e02] NICT(2258349005) Timer F( 10000 ms ) STARTED
> 337066:25:26.528 INF: [CID=0x0e02] >>> REGISTER sip:softswitch.mydomain.net SIP/2.0 DST: 85.xxx.x.85:5060:UDP SRC: 85.xxx.x.200:5060 enc=0 bytes=770
> 337066:25:26.529 DBG: [CID=0x0e02]
> 337066:25:26.529 DBG: [CID=0x0e02] REGISTER sip:softswitch.mydomain.net SIP/2.0
> 337066:25:26.529 DBG: [CID=0x0e02] From: "SoftClient01" <sip:862...@so...>;tag=6613195e
> 337066:25:26.529 DBG: [CID=0x0e02] To: "SoftClient01" <sip:862...@si...>
> 337066:25:26.529 DBG: [CID=0x0e02] Via: SIP/2.0/UDP 85.xxx.x.200:5060;iid=16143;branch=z9hG4bK4cba98df6938dd118b1d98e0e14f783f-69bc052885e284aa4b0d2044600ff220;uas-addr=85.xxx.x.85;rport
> 337066:25:26.529 DBG: [CID=0x0e02] Via: SIP/2.0/UDP 192.168.10.3:37934;branch=z9hG4bK-d87543-30153228e21d0017-1--d87543-;rport=37934;received=192.168.10.3
> 337066:25:26.529 DBG: [CID=0x0e02] CSeq: 1 REGISTER
> 337066:25:26.529 DBG: [CID=0x0e02] Call-ID: OWFkMTI4ODljOGQ3NjA1MDM1NzY0ZGViZDUzMjZmMmM.
> 337066:25:26.529 DBG: [CID=0x0e02] Contact: <sip:862...@85....x.200:5062;rinstance=6e999ef109459e0f>
> 337066:25:26.529 DBG: [CID=0x0e02] User-Agent: OpenSBC v1.1.5-6
> 337066:25:26.529 DBG: [CID=0x0e02] Expires: 3600
> 337066:25:26.529 DBG: [CID=0x0e02] Max-Forwards: 69
> 337066:25:26.529 DBG: [CID=0x0e02] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
> 337066:25:26.529 DBG: [CID=0x0e02] Content-Length: 0
> 337066:25:26.529 DBG: [CID=0x0e02]
> 337066:25:26.529 DBG: [CID=0x0e02]
> 337066:25:26.544 INF: [CID=0x0e02] <<< SIP/2.0 401 Unauthorized SRC: 85.xxx.x.85:5060:UDP enc=0 bytes=681
> 337066:25:26.545 DBG: [CID=0x0e02]
> 337066:25:26.545 DBG: [CID=0x0e02] SIP/2.0 401 Unauthorized
> 337066:25:26.545 DBG: [CID=0x0e02] From: "SoftClient01" <sip:862...@so...>;tag=6613195e
> 337066:25:26.545 DBG: [CID=0x0e02] To: "SoftClient01" <sip:862...@si...>;tag=1_1146_t930_7352
> 337066:25:26.545 DBG: [CID=0x0e02] Via: SIP/2.0/UDP 85.xxx.x.200:5060;iid=16143;branch=z9hG4bK4cba98df6938dd118b1d98e0e14f783f-69bc052885e284aa4b0d2044600ff220;uas-addr=85.xxx.x.85;rport=5060;received=85.xxx.x.200
> 337066:25:26.545 DBG: [CID=0x0e02] Via: SIP/2.0/UDP 192.168.10.3:37934;branch=z9hG4bK-d87543-30153228e21d0017-1--d87543-;rport=37934;received=192.168.10.3
> 337066:25:26.545 DBG: [CID=0x0e02] CSeq: 1 REGISTER
> 337066:25:26.545 DBG: [CID=0x0e02] Call-ID: OWFkMTI4ODljOGQ3NjA1MDM1NzY0ZGViZDUzMjZmMmM.
> 337066:25:26.545 DBG: [CID=0x0e02] WWW-Authenticate: Digest realm="ccsip", nonce="455432f8519b24380e57d40f5f2cca7f", algorithm=MD5, qop="auth"
> 337066:25:26.545 DBG: [CID=0x0e02] Content-Length: 0
> 337066:25:26.545 DBG: [CID=0x0e02]
> 337066:25:26.545 DBG: [CID=0x0e02]
> 337066:25:26.546 DBG: [CID=0x0e02] Finding transaction for SIP/2.0 401 Unauthorized
> 337066:25:26.546 DBG: [CID=0x0e02] Setting Transaction ID to OWFkMTI4ODljOGQ3NjA1MDM1NzY0ZGViZDUzMjZmMmM.|z9hG4bK4cba98df6938dd118b1d98e0e14f783f-69bc052885e284aa4b0d2044600ff220|REGISTER
> 337066:25:26.547 DTL: [CID=0x0e02] Found NICT|OWFkMTI4ODljOGQ3NjA1MDM1NzY0ZGViZDUzMjZmMmM.|z9hG4bK4cba98df6938dd118b1d98e0e14f783f-69bc052885e284aa4b0d2044600ff220|REGISTER for SIP/2.0 401 Unauthorized
> 337066:25:26.548 DTL: [CID=0x0e02] NICT(2258349005) Event(SIPMessage) - SIP/2.0 401 Unauthorized
> 337066:25:26.548 DBG: [CID=0x0e02] TRANSACTION: (NICT) SIP/2.0 401 Unauthorized State: 1
> 337066:25:26.548 DTL: [CID=0x0e02] NICT(2258349005) StateTrying->StateCompleted
> 337066:25:26.548 DBG: [CID=0x0e02] NICT(2258349005) Timer E STOPPED
> 337066:25:26.549 DBG: [CID=0x0e02] NICT(2258349005) Timer F STOPPED
> 337066:25:26.549 DBG: [CID=0x0e02] Event: SIPStack::Enqueue(SIP/2.0 401 Unauthorized)
> 337066:25:26.550 DBG: [CID=0x0e02] NICT(2258349005) Timer K( 5000 ms ) STARTED
> 337066:25:26.551 DBG: [CID=0x0e02] Event: B2BUserAgent::ProcessEvent( SIP/2.0 401 Unauthorized )
> 337066:25:26.553 DBG: [CID=0x0e02] Event: Setting UA Core [Proxy] to handle event REGISTER
> 337066:25:26.553 DTL: [CID=0x0e02] Event: ---> Inbound - SIP/2.0 401 Unauthorized
> 337066:25:26.554 DBG: [CID=0x0e02] *** MESSAGE ARRIVAL *** for SIP Session OWFkMTI4ODljOGQ3NjA1MDM1NzY0ZGViZDUzMjZmMmM.
> 337066:25:32.700 DBG: [CID=0x0e02] NICT(2258349005) Timer K( 5000 ms ) EXPIRED
> 337066:25:32.706 DBG: [CID=0x0e02] Finding transaction for SIP/2.0 401 Unauthorized
> 337066:25:32.707 DBG: [CID=0x0e02] Setting Transaction ID to OWFkMTI4ODljOGQ3NjA1MDM1NzY0ZGViZDUzMjZmMmM.|z9hG4bK4cba98df6938dd118b1d98e0e14f783f-69bc052885e284aa4b0d2044600ff220|REGISTER
> 337066:25:32.708 DBG: [CID=0x0e02]
> 337066:25:32.708 DBG: [CID=0x0e02] *** TRANSACTION DOES NOT EXIST ***
> 337066:25:32.708 DBG: [CID=0x0e02] Message: SIP/2.0 401 Unauthorized
> 337066:25:32.708 DBG: [CID=0x0e02] Call-Id: OWFkMTI4ODljOGQ3NjA1MDM1NzY0ZGViZDUzMjZmMmM.
> 337066:25:32.708 DBG: [CID=0x0e02]
> 337066:25:32.709 DTL: [CID=0x06cb] *** QUEUED FOR DELETION *** SIPSession: OWFkMTI4ODljOGQ3NjA1MDM1NzY0ZGViZDUzMjZmMmM.
> 337066:25:32.709 DBG: [CID=0x0000] GC: First Stale Object ProxySession
> 337066:25:32.722 INF: [CID=0x0e02] *** DESTROYED *** PROXY Session OWFkMTI4ODljOGQ3NjA1MDM1NzY0ZGViZDUzMjZmMmM.
> 337066:25:32.722 DBG: [CID=0x0e02] PROXY: Session DESTROYED
>
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>
> Has somebody seen such a behavior ? Is my expectation right that the SBC should forward the request and not close the proxy session immediately ?
>
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> Many thanks in advance.
>
>
>
> Andre
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From: prasad k. <pra...@gm...> - 2008-06-16 13:42:26
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Hi all , I am new to blog , so please help me in this issue. i need build a RTSP server and client which has to start with basics connection establishment and teardown .. process let me know if their is server to download and install and iam usind fedora as my os .... please ............. -- With regards Prasad |
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From: OpenSIPStack F. <ope...@op...> - 2008-06-13 12:17:46
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Hi all, I have been searching the forum for hours now but can not find any hint what I' m doing wrong. I' m running OpenSBC in B2B upper registration mode and would like to have my clients registering whith our existing softswitch. I have set up everything according to the tutorials and howto's which can be found in this forum. The client sends a register message to the SBC, the SBC acknowledges that there is an upper registration route and forwards the request to the softswitch. The softswitch reguires digest authentication and sends a 401 unauthorized back. I would expect now that this message is relayed back to the original client in order to continue with the authentication. For some reason, that never happens. The client is trying and trying to authenticate. Here is a small sippet from the log file: 337043:56:45.661 INF: [CID=0x0853] Routing REGISTER for URI sip:xxx...@si... 337043:56:45.663 PWL: [CID=0x0000] -->> From: sip:xxx...@si... Target: REGISTER sip:xxx...@sw...;domain=switch.def.de 337043:56:45.665 INF: [CID=0x0853] *** UPPER REGISTRATION ENABLE *** for sip:xxx...@si... 337043:56:45.680 INF: [CID=0x0853] >>> SIP/2.0 100 Trying DST: 10.70.4.9:5060:UDP SRC: 192.168.10.238:5060 enc=0 bytes=335 337043:56:45.689 INF: [CID=0x0853] >>> REGISTER sip:switch.def.de SIP/2.0 DST: 85.xxx.x.xx:5060:UDP SRC: 192.168.10.238:5060 enc=0 bytes=712 337043:56:45.702 INF: [CID=0x0853] <<< SIP/2.0 401 Unauthorized SRC: 85.xxx.x.xx:5060:UDP enc=0 bytes=653 337043:56:45.717 INF: [CID=0x0853] *** DESTROYED *** PROXY Session bbtzpwogsaqeoge@10.70.4.9 337043:56:46.110 INF: [CID=0x0853] <<< REGISTER sip:sip.ccgmbh.de SIP/2.0 SRC: 10.70.4.9:5060:UDP enc=0 bytes=469 ........ 337044:10:06.997 INF: [CID=0x0853] <<< REGISTER sip:sip.ccgmbh.de SIP/2.0 SRC: 10.70.4.9:5060:UDP enc=0 bytes=469 337044:10:07.005 INF: [CID=0x0853] >>> SIP/2.0 100 Trying DST: 10.70.4.9:5060:UDP SRC: 192.168.10.238:5060 enc=0 bytes=335 337044:10:11.013 INF: [CID=0x0853] <<< REGISTER sip:sip.ccgmbh.de SIP/2.0 SRC: 10.70.4.9:5060:UDP enc=0 bytes=469 337044:10:11.020 INF: [CID=0x0853] >>> SIP/2.0 100 Trying DST: 10.70.4.9:5060:UDP SRC: 192.168.10.238:5060 enc=0 bytes=335 337044:10:15.009 INF: [CID=0x0853] <<< REGISTER sip:sip.ccgmbh.de SIP/2.0 SRC: 10.70.4.9:5060:UDP enc=0 bytes=469 337044:10:15.017 INF: [CID=0x0853] >>> SIP/2.0 100 Trying DST: 10.70.4.9:5060:UDP SRC: 192.168.10.238:5060 enc=0 bytes=335 337044:10:19.005 INF: [CID=0x0853] <<< REGISTER sip:sip.ccgmbh.de SIP/2.0 SRC: 10.70.4.9:5060:UDP enc=0 bytes=469 337044:10:19.013 INF: [CID=0x0853] >>> SIP/2.0 100 Trying DST: 10.70.4.9:5060:UDP SRC: 192.168.10.238:5060 enc=0 bytes=335 337044:10:23.003 INF: [CID=0x0853] <<< REGISTER sip:sip.ccgmbh.de SIP/2.0 SRC: 10.70.4.9:5060:UDP enc=0 bytes=469 337044:10:23.011 INF: [CID=0x0853] >>> SIP/2.0 100 Trying DST: 10.70.4.9:5060:UDP SRC: 192.168.10.238:5060 enc=0 bytes=335 337044:10:26.999 INF: [CID=0x0853] <<< REGISTER sip:sip.ccgmbh.de SIP/2.0 SRC: 10.70.4.9:5060:UDP enc=0 bytes=469 337044:10:27.007 INF: [CID=0x0853] >>> SIP/2.0 100 Trying DST: 10.70.4.9:5060:UDP SRC: 192.168.10.238:5060 enc=0 bytes=335 337044:10:30.996 INF: [CID=0x0853] <<< REGISTER sip:sip.ccgmbh.de SIP/2.0 SRC: 10.70.4.9:5060:UDP enc=0 bytes=469 337044:10:31.003 INF: [CID=0x0853] >>> SIP/2.0 100 Trying DST: 10.70.4.9:5060:UDP SRC: 192.168.10.238:5060 enc=0 bytes=335 337044:10:34.992 INF: [CID=0x0853] <<< REGISTER sip:sip.ccgmbh.de SIP/2.0 SRC: 10.70.4.9:5060:UDP enc=0 bytes=469 337044:10:34.999 INF: [CID=0x0853] >>> SIP/2.0 100 Trying DST: 10.70.4.9:5060:UDP SRC: 192.168.10.238:5060 enc=0 bytes=335 337044:11:08.461 INF: [CID=0x0853] <<< REGISTER sip:sip.ccgmbh.de SIP/2.0 SRC: 10.70.4.9:5060:UDP enc=0 bytes=469 Has somebody seen such a behavior ? Is my expectation right that the SBC should forward the request and not close the proxy session immediately ? Many thanks in advance. |
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From: Thomas R. <tho...@it...> - 2008-06-13 11:37:58
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Hi. We are currently considering wether we should use OSS for a small client application (possibly using ATLSIP.dll). I've been wondering if it is possible / how difficult it is to change the buffer size because we need smaller delays (this is for LAN calls only not the internet). I've been recording delays of nearly 300ms which is probably ok through the internet, but is a problem for LAN connections at call centers,.. -- I did measure this by enabling recording on our PBX and then using a SIP phone [snom 360] on one end and a PC with Solegy phone / OSSPhone on the other end. At the pc I put a cable from microphone to headset jack (with a condensator and a resistor ;)) so that we can measure the delay properly by just having a look at the wave file. Please let me know if this is maybe just a parameter somewhere or not configurable at all. Regards, Mit freundlichen Grüßen Thomas Raschbacher ____________________________________________ itCampus Technology GmbH Österreich * Deutschland * Italien Dresdner Straße 45 /DG 1200 Wien tho...@it... Tel: +43 (1) 890 22 82 - 58 Fax: +43 (1) 890 22 82 - 958 http://www.itctec.com UID: ATU 6339 0618 Firmenbuchnr: FN292598t, Handelsgericht Wien Geschäftsführer: Andreas Günser, Andreas Lassmann Joint Venture von itCampus und MEC itCampus Gruppe Deutschland * Großbritannien * Italien * Österreich * Schweiz * Slowakei http://www.itcampus.eu |
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From: <jo...@op...> - 2008-06-12 23:18:49
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Hi Lucas, I do not have access to a Vista license currently. Perhaps the level 5 PWLIB and SIP logs would tell us something. Can you send them off-list? Joegen lucas martinez wrote: > Joegen, > Do you have any advise to help me working around on this matter. Thanks > > On Thu, Jun 5, 2008 at 8:34 AM, lucas martinez <mar...@gm...> wrote: > > >> Thanks for answer Joegen, im just wondering if this could help to avoid >> this kind of problem. Do you know how to solve this problem, a work around >> or what do i need to check? >> >> Thanks. >> >> >> On Wed, Jun 4, 2008 at 9:17 PM, Joegen E. Baclor <joe...@gm...> >> wrote: >> >> >>> Are you saying that the new pwlib with direct sound support has no >>> problem in vista? >>> >>> >>> lucas martinez wrote: >>> >>>> Hi, >>>> I´m using the ATLSIP.DLL and when i install the application in a PC with >>>> Windows Vista the audio is very bad, i have been looking for some >>>> information here and i found that seting SetSoundChannelBufferDepth( 10 >>>> >>> ) >>> >>>> instead of 5, but we still have the same problem. >>>> I found a new version of PWLIB(1.12.0) which support DirectSound, is >>>> >>> this >>> >>>> too hard to adapt, just wondering? >>>> >>>> Thank in advance. >>>> >>>> Lucas. >>>> >>>> >>> ------------------------------------------------------------------------- >>> >>>> Check out the new SourceForge.net Marketplace. >>>> It's the best place to buy or sell services for >>>> just about anything Open Source. >>>> http://sourceforge.net/services/buy/index.php >>>> _______________________________________________ >>>> opensipstack-devel mailing list >>>> ope...@li... >>>> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel >>>> >>>> >>>> >>>> >>> >>> ------------------------------------------------------------------------- >>> Check out the new SourceForge.net Marketplace. >>> It's the best place to buy or sell services for >>> just about anything Open Source. >>> http://sourceforge.net/services/buy/index.php >>> _______________________________________________ >>> opensipstack-devel mailing list >>> ope...@li... >>> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel >>> >>> >> > ------------------------------------------------------------------------- > Check out the new SourceForge.net Marketplace. > It's the best place to buy or sell services for > just about anything Open Source. > http://sourceforge.net/services/buy/index.php > _______________________________________________ > opensipstack-devel mailing list > ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel > > > > |
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From: lucas m. <mar...@gm...> - 2008-06-12 16:17:19
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Joegen, Do you have any advise to help me working around on this matter. Thanks On Thu, Jun 5, 2008 at 8:34 AM, lucas martinez <mar...@gm...> wrote: > Thanks for answer Joegen, im just wondering if this could help to avoid > this kind of problem. Do you know how to solve this problem, a work around > or what do i need to check? > > Thanks. > > > On Wed, Jun 4, 2008 at 9:17 PM, Joegen E. Baclor <joe...@gm...> > wrote: > >> Are you saying that the new pwlib with direct sound support has no >> problem in vista? >> >> >> lucas martinez wrote: >> > Hi, >> > I´m using the ATLSIP.DLL and when i install the application in a PC with >> > Windows Vista the audio is very bad, i have been looking for some >> > information here and i found that seting SetSoundChannelBufferDepth( 10 >> ) >> > instead of 5, but we still have the same problem. >> > I found a new version of PWLIB(1.12.0) which support DirectSound, is >> this >> > too hard to adapt, just wondering? >> > >> > Thank in advance. >> > >> > Lucas. >> > >> ------------------------------------------------------------------------- >> > Check out the new SourceForge.net Marketplace. >> > It's the best place to buy or sell services for >> > just about anything Open Source. >> > http://sourceforge.net/services/buy/index.php >> > _______________________________________________ >> > opensipstack-devel mailing list >> > ope...@li... >> > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel >> > >> > >> > >> >> >> >> ------------------------------------------------------------------------- >> Check out the new SourceForge.net Marketplace. >> It's the best place to buy or sell services for >> just about anything Open Source. >> http://sourceforge.net/services/buy/index.php >> _______________________________________________ >> opensipstack-devel mailing list >> ope...@li... >> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel >> > > |
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From: <jo...@op...> - 2008-06-08 02:59:22
|
Whit and H.Kropf, A leak in OpalFlexiTranscoder proxy class maybe causing this behavior. OpalFlexiTranscoder was first introduced by Ilian Pinzon so that re-invites with a media channel change my allow the codec to be changed in mid session. OpalFlexiTranscoder, however, does not have a destructor override. This allowed the OpalTranscoder object it created in the heap to be forever lost. I have committed a patch for this in CVS and is ready for testing. Let me know if if does/doesn't fix this issue. Thanks. Joegen Joegen E. Baclor wrote: > Ok this bug seems legitimate. I have created a ticket for this in > assembla. > > http://www.assembla.com/spaces/opensbc/tickets/21 . > > If there some more info you could send that might help crush this bug, > please send them in. As a backgrounder, the Voice Age G.729 codec > educational license only allows for one channel to be opened. This is > the reason why OPAL statically flag its usage. Thus, it is very > possible that this bug only became evident with G.729 but may also be > true for other codecs as well. The only diff is other codec might be > more forgiving than G.729. More information that would > confirm/disprove this case, please send them in. > > Thanks > > Joegen > > Whit Thiele wrote: > >> I have also been able to replicate this issue. So far I've only seen it >> happen using G.729. The first call seems to go through, but every subsequent >> call attempt fails. I am going to try recompiling an older snapshot of the >> CVS source I have. >> >> H.Kropf - Have you found out anything new on this issue? >> >> >> Whit >> >> -----Original Message----- >> From: ope...@li... >> [mailto:ope...@li...] On Behalf Of >> Joegen E. Baclor >> Sent: Thursday, May 22, 2008 9:14 PM >> To: H.Kropf >> Cc: ope...@li... >> Subject: Re: [OpenSIPStack] [SF] RTP fault >> >> Hi, >> >> Thanks for the logs. They were very helpful. It is evident that the >> previous call did not destroy the codec properly. I will try to >> replicate this on my system. Ilian is currently on leave so it might >> take a while for him to catch this thread. I have created a ticket for >> this: http://www.assembla.com/spaces/opensbc/tickets/20 >> >> Joegen >> >> H.Kropf wrote: >> >> >>> Hi >>> >>> Log (for pwlib - level 6) - in attachments >>> >>> >>> Joegen E. Baclor wrote: >>> >>> >>>> Hmmn. Strange. Seems like the codec from the previous call has not >>>> been destroyed. Can you send me a maximum level log of two >>>> consecutive calls right after fresh startup of your SoftPhone? >>>> >>>> H.Kropf wrote: >>>> >>>> >>>> >>>>> Hi >>>>> >>>>> >>>>> >>>>> >>>>>>> can you put a break-point in OpalTranscoder::ConvertFrames() and >>>>>>> figure out where exactly it fails? >>>>>>> >>>>>>> >>>>>>> >>>>> I can :) >>>>> >>>>> >>>>> bool OpalMediaPatch::Sink::WriteFrame(RTP_DataFrame & sourceFrame) >>>>> { >>>>> ...... >>>>> if (!primaryCodec->ConvertFrames(sourceFrame, intermediateFrames)) >>>>> { >>>>> PTRACE(1, "Patch\tMedia conversion (primary) failed"); >>>>> return false; >>>>> } >>>>> ...... >>>>> } >>>>> >>>>> BOOL OpalTranscoder::ConvertFrames(const RTP_DataFrame & input, >>>>> RTP_DataFrameList & output) >>>>> { >>>>> ....... >>>>> return Convert(input, output[0]); >>>>> } >>>>> >>>>> BOOL OpalFlexiTranscoder::Convert( const RTP_DataFrame & input, >>>>> RTP_DataFrame & output) >>>>> { >>>>> return m_Transcoder->Convert( input, output ); >>>>> } >>>>> >>>>> BOOL OpalFramedTranscoder::Convert(const RTP_DataFrame & input, >>>>> RTP_DataFrame & output) >>>>> { >>>>> .... >>>>> while (inputLength > 0) >>>>> { >>>>> ...... >>>>> if (!ConvertFrame(inputPtr, consumed, outputPtr, created)) >>>>> return FALSE; >>>>> ....... >>>>> } >>>>> ....... >>>>> } >>>>> >>>>> BOOL OpalFramedTranscoder::ConvertFrame(const BYTE * inputPtr, >>>>> PINDEX & /*consumed*/, BYTE * outputPtr, PINDEX & /*created*/) >>>>> { >>>>> return ConvertFrame(inputPtr, outputPtr); >>>>> } >>>>> >>>>> >>>>> >>>>> BOOL Opal_PCM_G729::ConvertFrame(const BYTE * src, BYTE * dst) >>>>> { >>>>> if (voiceAgeEncoderInUse != this) return FALSE; // >>>>> !!!!!!!!!!!! <<<=== this place >>>>> ...... >>>>> } >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>>> Hi, >>>>>> >>>>>> Are you a C++ developer? If so, can you put a break-point in >>>>>> OpalTranscoder::ConvertFrames() and figure out where exactly it fails? >>>>>> >>>>>> Joegen >>>>>> >>>>>> H.Kropf wrote: >>>>>> >>>>>> >>>>>> >>>>>>> Hello >>>>>>> >>>>>>> After a last update of library from CVS (2008-05-20), my softphone >>>>>>> (on OSS library) makes only one successful call after start. In >>>>>>> next calls there is no voice. In PTRACE-log there are many such >>>>>>> records >>>>>>> >>>>>>> Media Patch:1eb9410 PWL: [CID=0x0000] Patch Media conversion >>>>>>> (primary) failed >>>>>>> Media Patch:1eb9410 PWL: [CID=0x0000] Patch Media conversion >>>>>>> (primary) failed >>>>>>> Media Patch:1eb9410 PWL: [CID=0x0000] Patch Media conversion >>>>>>> (primary) failed >>>>>>> Media Patch:1eb9410 PWL: [CID=0x0000] Patch Media conversion >>>>>>> (primary) failed >>>>>>> >>>>>>> >>>>>>> In the previous version (CVS 2008-05-12) this problem did not exist >>>>>>> >>>>>>> >>>>>>> >>>>>>> >> ------------------------------------------------------------------------- >> >> >>>>>>> This SF.net email is sponsored by: Microsoft Defy all challenges. >>>>>>> Microsoft(R) Visual Studio 2008. >>>>>>> http://clk.atdmt.com/MRT/go/vse0120000070mrt/direct/01/ >>>>>>> _______________________________________________ >>>>>>> opensipstack-devel mailing list >>>>>>> ope...@li... >>>>>>> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >> ------------------------------------------------------------------------- >> >> >>>>>> This SF.net email is sponsored by: Microsoft Defy all challenges. >>>>>> Microsoft(R) Visual Studio 2008. >>>>>> http://clk.atdmt.com/MRT/go/vse0120000070mrt/direct/01/ >>>>>> _______________________________________________ >>>>>> opensipstack-devel mailing list >>>>>> ope...@li... >>>>>> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel >>>>>> >>>>>> >>>>>> >>>>>> >> ------------------------------------------------------------------------- >> >> >>>>> This SF.net email is sponsored by: Microsoft Defy all challenges. >>>>> Microsoft(R) Visual Studio 2008. >>>>> http://clk.atdmt.com/MRT/go/vse0120000070mrt/direct/01/ >>>>> _______________________________________________ >>>>> opensipstack-devel mailing list >>>>> ope...@li... >>>>> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel >>>>> >>>>> >>>>> >>>>> >>>>> >>>> ------------------------------------------------------------------------- >>>> >>>> >> >> >>>> This SF.net email is sponsored by: Microsoft Defy all challenges. >>>> Microsoft(R) Visual Studio 2008. >>>> http://clk.atdmt.com/MRT/go/vse0120000070mrt/direct/01/ >>>> _______________________________________________ >>>> opensipstack-devel mailing list >>>> ope...@li... >>>> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel >>>> >>>> >>>> >>>> >>> ------------------------------------------------------------------------ >>> >>> No virus found in this incoming message. >>> Checked by AVG. >>> Version: 7.5.524 / Virus Database: 269.23.21/1458 - Release Date: >>> >>> >> 5/21/2008 7:21 AM >> >> >> ------------------------------------------------------------------------- >> This SF.net email is sponsored by: Microsoft >> Defy all challenges. Microsoft(R) Visual Studio 2008. >> http://clk.atdmt.com/MRT/go/vse0120000070mrt/direct/01/ >> _______________________________________________ >> opensipstack-devel mailing list >> ope...@li... >> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel >> >> No virus found in this incoming message. >> Checked by AVG. >> Version: 8.0.100 / Virus Database: 269.23.21/1458 - Release Date: 5/21/2008 >> 7:21 AM >> >> >> >> >> > > > ------------------------------------------------------------------------- > Check out the new SourceForge.net Marketplace. > It's the best place to buy or sell services for > just about anything Open Source. > http://sourceforge.net/services/buy/index.php > _______________________________________________ > opensipstack-devel mailing list > ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel > > > > |
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From: lucas m. <mar...@gm...> - 2008-06-05 11:44:42
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Thanks for answer Joegen, im just wondering if this could help to avoid this kind of problem. Do you know how to solve this problem, a work around or what do i need to check? Thanks. On Wed, Jun 4, 2008 at 9:17 PM, Joegen E. Baclor <joe...@gm...> wrote: > Are you saying that the new pwlib with direct sound support has no > problem in vista? > > > lucas martinez wrote: > > Hi, > > I´m using the ATLSIP.DLL and when i install the application in a PC with > > Windows Vista the audio is very bad, i have been looking for some > > information here and i found that seting SetSoundChannelBufferDepth( 10 ) > > instead of 5, but we still have the same problem. > > I found a new version of PWLIB(1.12.0) which support DirectSound, is this > > too hard to adapt, just wondering? > > > > Thank in advance. > > > > Lucas. > > ------------------------------------------------------------------------- > > Check out the new SourceForge.net Marketplace. > > It's the best place to buy or sell services for > > just about anything Open Source. > > http://sourceforge.net/services/buy/index.php > > _______________________________________________ > > opensipstack-devel mailing list > > ope...@li... > > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel > > > > > > > > > > ------------------------------------------------------------------------- > Check out the new SourceForge.net Marketplace. > It's the best place to buy or sell services for > just about anything Open Source. > http://sourceforge.net/services/buy/index.php > _______________________________________________ > opensipstack-devel mailing list > ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel > |
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From: OpenSIPStack F. <ope...@op...> - 2008-06-05 10:31:30
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what are the unfinished modules in the open sbc |
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From: Joegen E. B. <joe...@gm...> - 2008-06-05 00:17:38
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Are you saying that the new pwlib with direct sound support has no problem in vista? lucas martinez wrote: > Hi, > I´m using the ATLSIP.DLL and when i install the application in a PC with > Windows Vista the audio is very bad, i have been looking for some > information here and i found that seting SetSoundChannelBufferDepth( 10 ) > instead of 5, but we still have the same problem. > I found a new version of PWLIB(1.12.0) which support DirectSound, is this > too hard to adapt, just wondering? > > Thank in advance. > > Lucas. > ------------------------------------------------------------------------- > Check out the new SourceForge.net Marketplace. > It's the best place to buy or sell services for > just about anything Open Source. > http://sourceforge.net/services/buy/index.php > _______________________________________________ > opensipstack-devel mailing list > ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel > > > |
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