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From: OpenSIPStack F. <ope...@op...> - 2008-07-22 18:47:44
|
Hi, With the Visual Studio build, if I don't want the build to automatically copy a bunch of files to \windows\system32 and modify the custom build steps to copy them to a local directory instead, short of adding this local directory to my PATH, how can I get configure to use a different search path to find the external dependencies? Thanks, -justin |
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From: OpenSIPStack F. <ope...@op...> - 2008-07-22 17:18:42
|
Hi thanks for the suggestions since FAR NAT have always problem when we do hosted places since we stay in india, its too far to host the applications there so we are building our own to get addressed the problem what we have described any documents for the same, Low level API, where can i look that is this development need to be done over PHP or XML ? any suggestions balaji |
|
From: OpenSIPStack F. <ope...@op...> - 2008-07-22 12:42:18
|
It is possible to use OSBC in front of SER as a registrar using upper registration. And, at the current time, OSBC can perform parallel forking, far end NAT, and act as a load balancer using only the configuration files. OSBC also writes call logs into a directory for each call. The other functionalities are not quite as straightforward # IVR - OSBC has a media server built-in, but there is no pre-built open-source app to play IVR, so this would need to be developed # Call Disconnect - This is possible using the low level API, but there is no pre-built open source app currently # LCR - This is possible using the low level API, but there is no pre-built open source app available currently # Conference Bridge - Not suppored Note that Solegy has created all of the apps that you refer to and offers them as a hosted service. More information is available here at the Solegy website. |
|
From: OpenSIPStack F. <ope...@op...> - 2008-07-22 09:31:46
|
Hi Iam having existing setup with SER but SER has billing issues, its not accurate so i would like to use OpensBC as billing router between my Voip provider and SER so users will register with SER, and ser send the calls to Opensbc opensbc should take care of serial forking and parellel forking with VoIP providers can this be done using opensSBC 1. far end NAT 2. call disconnect ( balance go negative) 3. loadbalance routes 4. LCR 5. IVR 6. Conference Bridge 7. CDR thanks balaji |
|
From: <jo...@op...> - 2008-07-21 04:32:23
|
Hi Othmar, TCP support in opensipstack is alpha code. It is not yet expected to work. Joegen oth...@in... wrote: > Hello, > > we are currently using the opensipstack and the openSBC for our > client/server development. > Both tools work fine except for TCP, > some SIP packets are bigger than 1300 bytes so TCP must be used according > to the RFCs. > > Q: Is this a known problem? > > In detail: > TCP sending (forwarding) is OK when the the packet was received via UDP > from a NATed client, > but receiving the response via TCP ends with 3 or more SIP packets (200 > OKs) merged in one. > It looks like this is a receiving and sliding TCP window problem. (??) > But in case the NATed client send an INVITE via TCP (about 1495 Bytes) > this is just ignored, > in wireshark I see that the packet was received in TCP/IP stack but it is > never seen in the SIPstack and no tracep output is generated. > > I am currently analyzing these problems. > But if you know already whats the problem it will be very helpfull to > inform me. > > (Test & Development Environment: > OpenSipStack 1.1.7, OpenSBC 1.1.4 > MS-Windows2000SP4+, > Microsoft Visual Studio 2005 (VC Express) > Version 8.0.50727.762 (SP.050727-7600) > Microsoft .NET Framework Version 2.0.50727) > > Thanks in advance and > best regards > Othmar Unger > > > > > ------------------------------------------------------------------------- > This SF.Net email is sponsored by the Moblin Your Move Developer's challenge > Build the coolest Linux based applications with Moblin SDK & win great prizes > Grand prize is a trip for two to an Open Source event anywhere in the world > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > _______________________________________________ > opensipstack-devel mailing list > ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel > > > No virus found in this incoming message. > Checked by AVG - http://www.avg.com > Version: 8.0.138 / Virus Database: 270.5.2/1561 - Release Date: 7/18/2008 6:35 PM > > > > |
|
From: <oth...@in...> - 2008-07-19 15:26:32
|
Hello, we are currently using the opensipstack and the openSBC for our client/server development. Both tools work fine except for TCP, some SIP packets are bigger than 1300 bytes so TCP must be used according to the RFCs. Q: Is this a known problem? In detail: TCP sending (forwarding) is OK when the the packet was received via UDP from a NATed client, but receiving the response via TCP ends with 3 or more SIP packets (200 OKs) merged in one. It looks like this is a receiving and sliding TCP window problem. (??) But in case the NATed client send an INVITE via TCP (about 1495 Bytes) this is just ignored, in wireshark I see that the packet was received in TCP/IP stack but it is never seen in the SIPstack and no tracep output is generated. I am currently analyzing these problems. But if you know already whats the problem it will be very helpfull to inform me. (Test & Development Environment: OpenSipStack 1.1.7, OpenSBC 1.1.4 MS-Windows2000SP4+, Microsoft Visual Studio 2005 (VC Express) Version 8.0.50727.762 (SP.050727-7600) Microsoft .NET Framework Version 2.0.50727) Thanks in advance and best regards Othmar Unger |
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From: Max G. <Max...@gm...> - 2008-07-19 08:02:09
|
Hi Joegen, its a pity. Thanks for your fast answer. Max -------- Original-Nachricht -------- > Datum: Sat, 19 Jul 2008 10:43:17 +0800 > Von: "jo...@op..." <joe...@gm...> > An: ope...@li... > Betreff: Re: [OpenSIPStack] TCP/TLS Support > Hi Max, > > TLS support is one of those stuffs that are badly needed but not needed > enough for OpenSBC functionality so it remained in the back burner until > now. Any takers? > > Joegen > > Max Groß wrote: > > Hi everyone, > > > > I'am looking for a sip library with TLS support and found OpenSipStack. > On your homepage I found a road map and there you said TCP/TLS Support for > version 1.6 is this up to date? > > > > Thanks for your answer. > > M. Gross > > > > > > ------------------------------------------------------------------------- > This SF.Net email is sponsored by the Moblin Your Move Developer's > challenge > Build the coolest Linux based applications with Moblin SDK & win great > prizes > Grand prize is a trip for two to an Open Source event anywhere in the > world > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > _______________________________________________ > opensipstack-devel mailing list > ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel -- GMX startet ShortView.de. Hier findest Du Leute mit Deinen Interessen! Jetzt dabei sein: http://www.shortview.de/wasistshortview.php?mc=sv_ext_mf@gmx |
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From: <jo...@op...> - 2008-07-19 02:43:20
|
Hi Max, TLS support is one of those stuffs that are badly needed but not needed enough for OpenSBC functionality so it remained in the back burner until now. Any takers? Joegen Max Groß wrote: > Hi everyone, > > I'am looking for a sip library with TLS support and found OpenSipStack. On your homepage I found a road map and there you said TCP/TLS Support for version 1.6 is this up to date? > > Thanks for your answer. > M. Gross > |
|
From: Max G. <Max...@gm...> - 2008-07-18 19:48:43
|
Hi everyone, I'am looking for a sip library with TLS support and found OpenSipStack. On your homepage I found a road map and there you said TCP/TLS Support for version 1.6 is this up to date? Thanks for your answer. M. Gross -- GMX startet ShortView.de. Hier findest Du Leute mit Deinen Interessen! Jetzt dabei sein: http://www.shortview.de/wasistshortview.php?mc=sv_ext_mf@gmx |
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From: OpenSIPStack F. <ope...@op...> - 2008-07-17 16:00:44
|
I do not see any harm done so I added it in CVS. However, m_ThreadPool will delete all threads when SIPUserAgent gets deleted. If they seem to be running forever, then it means SIPUserAgent is staying forever too. Deleting SIPUserAgent will delete the threads.
Joegen
> {quote:title=optotronic wrote:}{quote}
> I think I found one shutdown problem. In SIPUserAgent.cxx:
>
> {quote}void SIPUserAgent::Terminate()
> {
> GetTransportManager()->Terminate();
> m_SIPStack.Terminate( m_ThreadPool.GetSize() );
> m_ThreadPool.RemoveAll(); // <- added this
> }{quote}
>
> I added the last line to remove the workers from the thread pool. Otherwise, they appeared to be running forever.
>
> Does that look right, or is there somewhere else where they should be destroyed, that is apparently not working properly?
>
> I am still frequently getting hangs on shutdown, especially on multicore systems, but this seems to be an improvement. I'm using OSSPhone (MFC) for testing, as it seems to have the closest behavior to my Delphi test app.
>
> Finest regards,
> Bill Root
|
|
From: OpenSIPStack F. <ope...@op...> - 2008-07-17 15:46:17
|
I think I found one shutdown problem. In SIPUserAgent.cxx:
{quote}void SIPUserAgent::Terminate()
{
GetTransportManager()->Terminate();
m_SIPStack.Terminate( m_ThreadPool.GetSize() );
m_ThreadPool.RemoveAll(); // <- added this
}{quote}
I added the last line to remove the workers from the thread pool. Otherwise, they appeared to be running forever.
Does that look right, or is there somewhere else where they should be destroyed, that is apparently not working properly?
I am still frequently getting hangs on shutdown, especially on multicore systems, but this seems to be an improvement. I'm using OSSPhone (MFC) for testing, as it seems to have the closest behavior to my Delphi test app.
Finest regards,
Bill Root
|
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From: <jo...@op...> - 2008-07-11 13:22:58
|
Aside from route fail-over and round-robin, OpenSBC do not offer service the advance class 4 softswitch can do. I am currently working on the open source version of the Solegy Service PDQ client. If you have the latest CVS copy you might already have noticed the new Solegy* files in OpenSBC source tree. This will enable OpenSBC users to get an account from Solegy for class 4 routing and billing. bay...@ic... wrote: > Dear All, > > I have successfully tried the opensbc class 5 feature by registering > some SIP subscribers and doing trunking to our class 4 softswitch. > > But I need to know whether opensbc could do class 4 feature to receive > VoIP traffic and then route the traffic to our class 4 softswitch. > > Please kindly confirm to me about the opensbc class 4 routing features > and please also direct me on how to configure the opensbc 1.1.4. > > I am looking forward to Your further response. Thank You very much at advance. > > Best Regards, > > Bayu Sukmanto > > ---------------------------------------------------------------- > Indonesia Comnet Plus Webmail > > > ------------------------------------------------------------------------- > Sponsored by: SourceForge.net Community Choice Awards: VOTE NOW! > Studies have shown that voting for your favorite open source project, > along with a healthy diet, reduces your potential for chronic lameness > and boredom. Vote Now at http://www.sourceforge.net/community/cca08 > _______________________________________________ > opensipstack-devel mailing list > ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel > > > > |
|
From: <bay...@ic...> - 2008-07-11 10:54:58
|
Dear All, I have successfully tried the opensbc class 5 feature by registering some SIP subscribers and doing trunking to our class 4 softswitch. But I need to know whether opensbc could do class 4 feature to receive VoIP traffic and then route the traffic to our class 4 softswitch. Please kindly confirm to me about the opensbc class 4 routing features and please also direct me on how to configure the opensbc 1.1.4. I am looking forward to Your further response. Thank You very much at advance. Best Regards, Bayu Sukmanto ---------------------------------------------------------------- Indonesia Comnet Plus Webmail |
|
From: James B. <jam...@gm...> - 2008-07-11 08:34:08
|
Hi Joegen, I've tried the latest code from CVS and every SUBSCRIBE and NOTIFY makes it through - just perfect. Thanks so much for such a fast resolution to this query. I'm very impressed. James On Fri, Jul 11, 2008 at 3:30 AM, Joegen E. Baclor <joe...@gm...> wrote: > Hi James, > > Please try the latest from CVS and let me know if you still experience the > race condition between overlapped transactions. The changes are mostly in > the OpenSIPStack library so do not forget to update your copy as well. > > Joegen > > James Brennan wrote: >> >> Hi Joegen, >> >> Thanks for the reply. >> Later NOTIFY messages sometimes make it out to the client >> and the client can receive and make calls after this occurs. >> It is only on startup of the client when the SUBSCRIBE and >> NOTIFY messages occur and later periodic NOTIFY updates >> by the Asterisk server when I see OpenSBC/SIPStack silently >> discard some NOTIFY messages. >> Hopefully this has answered the question you had about the FSM. >> >> James >> >> On Tue, Jul 8, 2008 at 1:27 PM, Joegen E. Baclor >> <joe...@gm...> wrote: >> >>> >>> Hi James, >>> >>> This is more of a SIP Stack question than and OpenSBC subject. It >>> seems you are hitting a race condition. I need to get back to you on >>> this after I've done some testing of my own. For the meantime, can you >>> confirm that the FSM has not dead-locked and is still accepting/relaying >>> transactions after this occurs? >>> >>> Joegen >>> >>> James Brennan wrote: >>> >>>> >>>> Hi, >>>> >>>> Sorry, not sure if this is an OpenSBC or OpenSIPStack query. >>>> >>>> I'm running OpenSBC (1.1.5-7 and OpenSIPStack 1.1.8). >>>> OpenSBC is in full-mode. >>>> >>>> When the client SUBSCRIBEs to presence of another user, >>>> the SIP server issues the 202 for the SUBSCRIBE and also an >>>> immediate NOTIFY within the same dialog. >>>> >>>> When these pass through OpenSBC, some times the NOTIFY >>>> will not make it through to the client. >>>> >>>> My client has 3 contacts and subscribes to presence of all 3. >>>> I've seen none, 1 or maybe 2 of these NOTIFY messages make >>>> it to the client. >>>> >>>> Having looked around the code and trace of OpenSIPStack >>>> it looks like it happens when the 202 for the SUBSCRIBE >>>> is processed, and the DestroySession occurs between processing >>>> of the NOTIFY. >>>> >>>> Perhaps something is wrong with my setup. >>>> Does this scenario sound like it should work!? >>>> >>>> I'll attach the trace from OpenSBC which captured >>>> the first 2 NOTIFY messages making it back to the client >>>> but the 3rd NOTIFY message did not. >>>> >>>> Any advice much appreciated, >>>> Thanks, >>>> James >>>> >>>> >>>> No virus found in this incoming message. >>>> Checked by AVG - http://www.avg.com >>>> Version: 8.0.138 / Virus Database: 270.4.5/1537 - Release Date: 7/6/2008 >>>> 5:26 AM >>>> >>>> >>>> ------------------------------------------------------------------------ >>>> >>>> >>>> ------------------------------------------------------------------------- >>>> Sponsored by: SourceForge.net Community Choice Awards: VOTE NOW! >>>> Studies have shown that voting for your favorite open source project, >>>> along with a healthy diet, reduces your potential for chronic lameness >>>> and boredom. Vote Now at http://www.sourceforge.net/community/cca08 >>>> ------------------------------------------------------------------------ >>>> >>>> _______________________________________________ >>>> Opensipstack-osbcdevel mailing list >>>> Ope...@li... >>>> https://lists.sourceforge.net/lists/listinfo/opensipstack-osbcdevel >>>> >>>> >>> >>> ------------------------------------------------------------------------- >>> Sponsored by: SourceForge.net Community Choice Awards: VOTE NOW! >>> Studies have shown that voting for your favorite open source project, >>> along with a healthy diet, reduces your potential for chronic lameness >>> and boredom. Vote Now at http://www.sourceforge.net/community/cca08 >>> _______________________________________________ >>> Opensipstack-osbcdevel mailing list >>> Ope...@li... >>> https://lists.sourceforge.net/lists/listinfo/opensipstack-osbcdevel >>> >>> >> >> No virus found in this incoming message. >> Checked by AVG - http://www.avg.com Version: 8.0.138 / Virus Database: >> 270.4.6/1539 - Release Date: 7/7/2008 6:35 PM >> >> >> >> > > |
|
From: Joegen E. B. <joe...@gm...> - 2008-07-11 02:30:50
|
Hi James, Please try the latest from CVS and let me know if you still experience the race condition between overlapped transactions. The changes are mostly in the OpenSIPStack library so do not forget to update your copy as well. Joegen James Brennan wrote: > Hi Joegen, > > Thanks for the reply. > Later NOTIFY messages sometimes make it out to the client > and the client can receive and make calls after this occurs. > It is only on startup of the client when the SUBSCRIBE and > NOTIFY messages occur and later periodic NOTIFY updates > by the Asterisk server when I see OpenSBC/SIPStack silently > discard some NOTIFY messages. > Hopefully this has answered the question you had about the FSM. > > James > > On Tue, Jul 8, 2008 at 1:27 PM, Joegen E. Baclor > <joe...@gm...> wrote: > >> Hi James, >> >> This is more of a SIP Stack question than and OpenSBC subject. It >> seems you are hitting a race condition. I need to get back to you on >> this after I've done some testing of my own. For the meantime, can you >> confirm that the FSM has not dead-locked and is still accepting/relaying >> transactions after this occurs? >> >> Joegen >> >> James Brennan wrote: >> >>> Hi, >>> >>> Sorry, not sure if this is an OpenSBC or OpenSIPStack query. >>> >>> I'm running OpenSBC (1.1.5-7 and OpenSIPStack 1.1.8). >>> OpenSBC is in full-mode. >>> >>> When the client SUBSCRIBEs to presence of another user, >>> the SIP server issues the 202 for the SUBSCRIBE and also an >>> immediate NOTIFY within the same dialog. >>> >>> When these pass through OpenSBC, some times the NOTIFY >>> will not make it through to the client. >>> >>> My client has 3 contacts and subscribes to presence of all 3. >>> I've seen none, 1 or maybe 2 of these NOTIFY messages make >>> it to the client. >>> >>> Having looked around the code and trace of OpenSIPStack >>> it looks like it happens when the 202 for the SUBSCRIBE >>> is processed, and the DestroySession occurs between processing >>> of the NOTIFY. >>> >>> Perhaps something is wrong with my setup. >>> Does this scenario sound like it should work!? >>> >>> I'll attach the trace from OpenSBC which captured >>> the first 2 NOTIFY messages making it back to the client >>> but the 3rd NOTIFY message did not. >>> >>> Any advice much appreciated, >>> Thanks, >>> James >>> >>> >>> No virus found in this incoming message. >>> Checked by AVG - http://www.avg.com >>> Version: 8.0.138 / Virus Database: 270.4.5/1537 - Release Date: 7/6/2008 5:26 AM >>> >>> >>> ------------------------------------------------------------------------ >>> >>> ------------------------------------------------------------------------- >>> Sponsored by: SourceForge.net Community Choice Awards: VOTE NOW! >>> Studies have shown that voting for your favorite open source project, >>> along with a healthy diet, reduces your potential for chronic lameness >>> and boredom. Vote Now at http://www.sourceforge.net/community/cca08 >>> ------------------------------------------------------------------------ >>> >>> _______________________________________________ >>> Opensipstack-osbcdevel mailing list >>> Ope...@li... >>> https://lists.sourceforge.net/lists/listinfo/opensipstack-osbcdevel >>> >>> >> ------------------------------------------------------------------------- >> Sponsored by: SourceForge.net Community Choice Awards: VOTE NOW! >> Studies have shown that voting for your favorite open source project, >> along with a healthy diet, reduces your potential for chronic lameness >> and boredom. Vote Now at http://www.sourceforge.net/community/cca08 >> _______________________________________________ >> Opensipstack-osbcdevel mailing list >> Ope...@li... >> https://lists.sourceforge.net/lists/listinfo/opensipstack-osbcdevel >> >> > > No virus found in this incoming message. > Checked by AVG - http://www.avg.com > Version: 8.0.138 / Virus Database: 270.4.6/1539 - Release Date: 7/7/2008 6:35 PM > > > > |
|
From: Christian W. <cwa...@gm...> - 2008-07-04 10:50:03
|
Hi, I also try to do the same and also have the same problems, maybe we doing something together: The easiest thing would be to have a library which provides VoIP via CAPI (like XCAPI) I already have the capi2032.dll running without VoIP. It can pass the data between different controller and already does PLCI and controller mapping... The next step is to provide Line interconnect (you need to mix the data) Which also needs media conversion (alaw (or whatsoever) 2 pcm 2 alaw) because the only way I know to mix is on PCM data (also DTMF tone recognition)... Kind regards Christian -----Ursprüngliche Nachricht----- Von: ope...@li... [mailto:ope...@li...] Im Auftrag von OpenSIPStack Forum Gesendet: Freitag, 4. Juli 2008 11:50 An: ope...@li... Betreff: [OpenSIPStack] Call transfer Hi, our firm produces ISDN Pbx under WindowsXP embedded (CPU: Intel Centrino 1,8 GHz, RAM: 1GB). The application is written in C++ and uses MFC library. Our requirements are: - we need to call a Sip phone connected to our LAN . - when the phone answer we want to transfer the voice packets received from it to an isdn line and vice versa, we want to transfer the voice packets received from the isdn line to the sip phone. - we can have, at the same time, up to 10 call tranfers of this type. In practice we are looking for a SIP library that allow us to: - make a call to a sip phone; - be notified about call status changes; - get the voice packets received from the sip phone; - send voice packets to the sip phone. We are trying to use OenSipStack but we have some problems. The main is how to get the packets received from the sip phone. We don't want the packets be sent to the sound board but to our application. At the same time we want to send to the sip phone our packets, not the ones comig from the microphone. Can anyone help us? Thank you ------------------------------------------------------------------------- Sponsored by: SourceForge.net Community Choice Awards: VOTE NOW! Studies have shown that voting for your favorite open source project, along with a healthy diet, reduces your potential for chronic lameness and boredom. Vote Now at http://www.sourceforge.net/community/cca08 _______________________________________________ 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: 8.0.101 / Virus Database: 270.4.4/1532 - Release Date: 03.07.2008 08:32 |
|
From: OpenSIPStack F. <ope...@op...> - 2008-07-04 10:42:45
|
Subclass PSoundChannel and implement Open, Read, Write, etc functions accordingly to fetch data from your ISDN line. Then override the method below and return an instance of your custom sound channel object.
PSoundChannel * OpalPCSSEndPoint::CreateSoundChannel(const OpalPCSSConnection & connection,
const OpalMediaFormat & mediaFormat,
BOOL isSource)
HTH,
Joegen
> {quote:title=microtek wrote:}{quote}
>
> Hi,
>
>
> our firm produces ISDN Pbx under WindowsXP embedded (CPU: Intel Centrino 1,8 GHz, RAM: 1GB).
> The application is written in C++ and uses MFC library.
>
> Our requirements are:
>
> - we need to call a Sip phone connected to our LAN .
> - when the phone answer we want to transfer the voice packets received from it to an isdn line and vice versa, we want to transfer the voice packets received from the isdn line to the sip phone.
> - we can have, at the same time, up to 10 call tranfers of this type.
>
> In practice we are looking for a SIP library that allow us to:
> - make a call to a sip phone;
> - be notified about call status changes;
> - get the voice packets received from the sip phone;
> - send voice packets to the sip phone.
>
>
> We are trying to use OenSipStack but we have some problems. The main is how to get the packets received from the sip phone. We don't want the packets be sent to the sound board but to our application. At the same time we want to send to the sip phone our packets, not the ones comig from the microphone.
>
>
> Can anyone help us?
>
> Thank you
>
>
|
|
From: OpenSIPStack F. <ope...@op...> - 2008-07-04 09:50:12
|
Hi, our firm produces ISDN Pbx under WindowsXP embedded (CPU: Intel Centrino 1,8 GHz, RAM: 1GB). The application is written in C++ and uses MFC library. Our requirements are: - we need to call a Sip phone connected to our LAN . - when the phone answer we want to transfer the voice packets received from it to an isdn line and vice versa, we want to transfer the voice packets received from the isdn line to the sip phone. - we can have, at the same time, up to 10 call tranfers of this type. In practice we are looking for a SIP library that allow us to: - make a call to a sip phone; - be notified about call status changes; - get the voice packets received from the sip phone; - send voice packets to the sip phone. We are trying to use OenSipStack but we have some problems. The main is how to get the packets received from the sip phone. We don't want the packets be sent to the sound board but to our application. At the same time we want to send to the sip phone our packets, not the ones comig from the microphone. Can anyone help us? Thank you |
|
From: <jo...@op...> - 2008-07-04 02:14:19
|
Do you know what particularly caused STUN to result to a forbidden from the provider? Christian Wallukat wrote: > Hi, > > > Did you set the STUN settings ? > I had the same problem if stun was active. > > > > Kind regards > > > Christian > > -----Ursprüngliche Nachricht----- > Von: ope...@li... > [mailto:ope...@li...] Im Auftrag von > haripriya alapati > Gesendet: Donnerstag, 3. Juli 2008 11:02 > An: ope...@li... > Betreff: [OpenSIPStack] [open sipstack] 403 Forbidden > > > Hi, > I am trying to call a tollfree number from OSSPhone. > OSSPhone is successfully registered. But call is disconnecting becoz of 403 > Forbidden. Can any one help me. > Here i am sending the log messages. > > >>>> REGISTER sip:202.71.134.13 SIP/2.0 DST: 202.71.134.13:5060:UDP SRC: >>>> > 172.16.0.5:5060 enc=0 bytes=785 > ----------------4:00:50.808---------------- > <<< SIP/2.0 100 Trying SRC: 202.71.134.13:5060:UDP enc=0 bytes=316 > ----------------4:00:50.821---------------- > <<< SIP/2.0 200 OK SRC: 202.71.134.13:5060:UDP enc=0 bytes=403 > >>>> INVITE sip:17187773456@202.71.134.13 SIP/2.0 DST: 202.71.134.13:5060:UDP >>>> > SRC: 172.16.0.5:5060 enc=0 bytes=1030 > ----------------4:01:00.943---------------- > <<< SIP/2.0 100 Trying SRC: 202.71.134.13:5060:UDP enc=0 bytes=305 > ----------------4:01:00.984---------------- > <<< SIP/2.0 403 Forbidden SRC: 202.71.134.13:5060:UDP enc=0 bytes=308 > ----------------4:01:00.999---------------- > >>>> ACK sip:17187773456@202.71.134.13 SIP/2.0 DST: 202.71.134.13:5060:UDP >>>> > SRC: 172.16.0.5:5060 enc=0 bytes=771 > > > Thanks in Advance. > > Regards, > Haripriya. > ------------------------------------------------------------------------- > Sponsored by: SourceForge.net Community Choice Awards: VOTE NOW! > Studies have shown that voting for your favorite open source project, > along with a healthy diet, reduces your potential for chronic lameness > and boredom. Vote Now at http://www.sourceforge.net/community/cca08 > _______________________________________________ > 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: 8.0.101 / Virus Database: 270.4.4/1530 - Release Date: 02.07.2008 > 08:05 > > > ------------------------------------------------------------------------- > Sponsored by: SourceForge.net Community Choice Awards: VOTE NOW! > Studies have shown that voting for your favorite open source project, > along with a healthy diet, reduces your potential for chronic lameness > and boredom. Vote Now at http://www.sourceforge.net/community/cca08 > _______________________________________________ > 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: 8.0.134 / Virus Database: 270.4.3/1529 - Release Date: 7/1/2008 7:23 PM > |
|
From: Christian W. <cwa...@gm...> - 2008-07-03 09:40:05
|
Hi, Did you set the STUN settings ? I had the same problem if stun was active. Kind regards Christian -----Ursprüngliche Nachricht----- Von: ope...@li... [mailto:ope...@li...] Im Auftrag von haripriya alapati Gesendet: Donnerstag, 3. Juli 2008 11:02 An: ope...@li... Betreff: [OpenSIPStack] [open sipstack] 403 Forbidden Hi, I am trying to call a tollfree number from OSSPhone. OSSPhone is successfully registered. But call is disconnecting becoz of 403 Forbidden. Can any one help me. Here i am sending the log messages. >>> REGISTER sip:202.71.134.13 SIP/2.0 DST: 202.71.134.13:5060:UDP SRC: 172.16.0.5:5060 enc=0 bytes=785 ----------------4:00:50.808---------------- <<< SIP/2.0 100 Trying SRC: 202.71.134.13:5060:UDP enc=0 bytes=316 ----------------4:00:50.821---------------- <<< SIP/2.0 200 OK SRC: 202.71.134.13:5060:UDP enc=0 bytes=403 >>> INVITE sip:17187773456@202.71.134.13 SIP/2.0 DST: 202.71.134.13:5060:UDP SRC: 172.16.0.5:5060 enc=0 bytes=1030 ----------------4:01:00.943---------------- <<< SIP/2.0 100 Trying SRC: 202.71.134.13:5060:UDP enc=0 bytes=305 ----------------4:01:00.984---------------- <<< SIP/2.0 403 Forbidden SRC: 202.71.134.13:5060:UDP enc=0 bytes=308 ----------------4:01:00.999---------------- >>> ACK sip:17187773456@202.71.134.13 SIP/2.0 DST: 202.71.134.13:5060:UDP SRC: 172.16.0.5:5060 enc=0 bytes=771 Thanks in Advance. Regards, Haripriya. ------------------------------------------------------------------------- Sponsored by: SourceForge.net Community Choice Awards: VOTE NOW! Studies have shown that voting for your favorite open source project, along with a healthy diet, reduces your potential for chronic lameness and boredom. Vote Now at http://www.sourceforge.net/community/cca08 _______________________________________________ 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: 8.0.101 / Virus Database: 270.4.4/1530 - Release Date: 02.07.2008 08:05 |
|
From: haripriya a. <har...@re...> - 2008-07-03 09:01:08
|
Hi, I am trying to call a tollfree number from OSSPhone. OSSPhone is successfully registered. But call is disconnecting becoz of 403 Forbidden. Can any one help me. Here i am sending the log messages. >>> REGISTER sip:202.71.134.13 SIP/2.0 DST: 202.71.134.13:5060:UDP SRC: 172.16.0.5:5060 enc=0 bytes=785 ----------------4:00:50.808---------------- <<< SIP/2.0 100 Trying SRC: 202.71.134.13:5060:UDP enc=0 bytes=316 ----------------4:00:50.821---------------- <<< SIP/2.0 200 OK SRC: 202.71.134.13:5060:UDP enc=0 bytes=403 >>> INVITE sip:17187773456@202.71.134.13 SIP/2.0 DST: 202.71.134.13:5060:UDP SRC: 172.16.0.5:5060 enc=0 bytes=1030 ----------------4:01:00.943---------------- <<< SIP/2.0 100 Trying SRC: 202.71.134.13:5060:UDP enc=0 bytes=305 ----------------4:01:00.984---------------- <<< SIP/2.0 403 Forbidden SRC: 202.71.134.13:5060:UDP enc=0 bytes=308 ----------------4:01:00.999---------------- >>> ACK sip:17187773456@202.71.134.13 SIP/2.0 DST: 202.71.134.13:5060:UDP SRC: 172.16.0.5:5060 enc=0 bytes=771 Thanks in Advance. Regards, Haripriya. |
|
From: Joegen E. B. <joe...@gm...> - 2008-07-01 06:23:11
|
Hi Gustavo,
I have re-instated PMEMORY_CHECK enabling for release builds. Just
define PMEMORY_CHECK=1. Let me know if you find out something
Joegen
jo...@op... wrote:
> Weird indeed. I will create a new status page that dumps memory
> snap-shot via the admin pages. I will keep you posted.
>
> Joegen
>
> Gustavo Curetti wrote:
>
>> Hi Joegen
>>
>> I found something really weird. If I change the order of headers in SIPMessage::Cleanup() the memory leak varies.
>> For example, if I clean m_AllowList and m_ViaList first of all the memory leak decrease:
>>
>> void SIPMessage::Cleanup(){ GlobalLock();
>>
>> if( m_AllowList != NULL ){ delete m_AllowList; m_AllowList = NULL; } /// special headers if( m_ViaList != NULL ){ delete m_ViaList; m_ViaList = NULL; }
>> .....
>>
>> I will continue searching. ¿Is there a way to configure PMEMORY_CHECK for release?
>>
>> Thanks for your help.
>>
>> Gustavo Curetti
>>
>>
>> From: cur...@ho...To: jo...@op...; joe...@gm...; ope...@li...Subject: RE: [OpenSIPStack] FW: Memory Leak in Proxy and Full ModeDate: Mon, 23 Jun 2008 21:42:45 +0200
>>
>>
>> Hi Joegen I think this is a memory leak because after all messages are sent the memory never go down.The code added in B2BUserAgent::Registrar::ProcessUpperRegKeepAlive() waits 5 seconds and sends the first Invite, then waits 5 seconds , sends the second Invite, waits 5 seconds more and finally sends 4998 Invites with a 40 ms delay between each one. void B2BUserAgent::Registrar::ProcessUpperRegKeepAlive(){ DWORD delay = 5000; DWORD numberOfPackets = 0; while( !m_UpperRegSync.Wait( delay ) ) { if(numberOfPackets < 5000) { 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: " + OString(numberOfPackets) + "@192.168.0.10\r\nMax-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 ) ); if (numberOfPackets > 1) delay = 40; numberOfPackets++; } }} The SIP timers B and H are set to 20 ms. #define SIP_TIMER_B 20 #define SIP_TIMER_H 20 After all messages are sent, I wait for a long time but the memory used doesn't decrease. I tried different settings: Case 1 5 sec -> 1 Invite -> 5 sec -> 1 Invite -> 5 sec -> 4998 Invite (40 ms) -> 5 min1)Mem Usage 8332k 8940k 9080k 9400k2)Mem Usage 8364k 8952k 8976k 9416k3)Mem Usage 8300k 8888k 8908k 9368k Case 2 5 sec -> 1 Invite -> 5 sec -> 1 Invite -> 5 sec -> 9998 Invite (40 ms) -> 5 min1)Mem Usage 8308k 8900k 9000k 9696k2)Mem Usage 8320k 8908k 8928k 9760k The leak is bigger when the amount of messages are higher.Case 3 5 sec -> 1 Invite -> 5 sec -> 1 Invite -> 5 sec -> 4998 Invite (30 ms) -> 5 min1)Mem Usage 8252k 8832k 8852k 9532k 2)Mem Usage 8256k 8844k 8872k 9536k Case 4 5 sec -> 1 Invite -> 5 sec -> 1 Invite -> 5 sec -> 4998 Invite (20 ms) -> 5 min1)Mem Usage 8276k 8864k 8884k 9728k 2)Mem Usage 8240k 8828k 8848k 9728k The leak is bigger when the delay is shorter. Case 5 5 sec -> 1 Invite -> 5 sec -> 1 Invite -> 5 sec -> 4998 Invite (80 ms) -> 5 min1)Mem Usage 8344k 8932k 8952k 9332k2)Mem Usage 8376k 8968k 8992k 9276k The only type of error in the log is : 102:42:36.691 ERR: [CID=0x0000] UDP Socket Read Error (Socket not connected) in case 1 appears 5219 times. I don't know if this error is linked with the leak. The log is attached. This behavior doesn't seem to be because of transactions waiting for timers. Maybe is because some type of conflict with shared resources or mutexs, but I really don't know. Any idea? If you need more info, please let me know. One more thing, I suggest to increase MAX_SIP_MESSAGE_LENGTH because i had problems with Invites with a lot of headers (record-routes, vias, etc) and long bodies. #define MAX_SIP_MESSAGE_LENGTH 2048 ----> #define MAX_SIP_MESSAGE_LENGTH 4000 Thanks for your help. Gustavo Curetti
>>
>>
>>
>>> Date: Sat, 21 Jun 2008 10:17:22 +0800> To: cur...@gm...> From: joe...@gm...> CC: joe...@gm...; ope...@li...> Subject: Re: [OpenSIPStack] FW: Memory Leak in Proxy and Full Mode> > Hi Gustavo,> > This time, I am not sure if what you are seeing in perfmon is actually a > leak or simply transactions waiting for timers to fire so it could > transition from Completed to Terminated state. I tried to fire the same > in my perfmon and the private mem leveled at around 80 in the graph. > Same place where yours leveled to a straight line. Can you share more > info why you think this is a leak?> > Joegen> > > Gustavo Curetti wrote:> > Hi Joegen,> > > > Another case of memory leak is attached. (no debugging mode).> > I send 5000 messages with 40 ms delay between each message.> > A jpg of the perfmon is attached too. (Private Bytes)> > > > Thanks for your help> >> > Gustavo Curetti> >> >> > ------------------------------------------------------------------------> >> > > Date: Fri, 20 Jun 2008 09:33:12 +0800> > > To: cur...@gm...; ope...@li...> > > From: joe...@gm...> > > CC: joe...@gm...> > > Subject: Re: [OpenSIPStack] FW: Memory Leak in Proxy and Full Mode> > >> > > Right! Patch is in CVS.> > >> > > Joegen> > >> > > Gustavo Curetti wrote:> > > > Hi Joegen,> > > >> > > > Thanks for your help. The other header which is missing in the > > SIPMessage::CleanUp() is the m_SessionExpires ("session-expires"). I > > will continue with my tests and let you know.> > > >> > > > Gustavo> > > >> > > >> > > >> > >> > >> > > > > -------------------------------------------------------------------------> > > 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> >> >> > ------------------------------------------------------------------------> > ¿Aburrido? Ingresá ya y divertite como nunca en MSN Juegos. MSN Juegos > > <http://juegos.ar.msn.com/>> > ------------------------------------------------------------------------> >> > 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
>>>
>>>
>> Descargá GRATIS el poder del nuevo Internet Explorer 7. Internet Explorer 7
>> _________________________________________________________________
>> Ingresá ya a MSN en Concierto y disfrutá los recitales en vivo de tus artistas favoritos.
>> http://msninconcert.msn.com/music/archive/es-la/archive.aspx
>> -------------------------------------------------------------------------
>> 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
> ------------------------------------------------------------------------
>
>
> No virus found in this incoming message.
> Checked by AVG.
> Version: 8.0.101 / Virus Database: 270.4.2/1523 - Release Date: 6/28/2008 7:00 AM
>
|
|
From: <jo...@op...> - 2008-06-28 03:17:08
|
Weird indeed. I will create a new status page that dumps memory
snap-shot via the admin pages. I will keep you posted.
Joegen
Gustavo Curetti wrote:
> Hi Joegen
>
> I found something really weird. If I change the order of headers in SIPMessage::Cleanup() the memory leak varies.
> For example, if I clean m_AllowList and m_ViaList first of all the memory leak decrease:
>
> void SIPMessage::Cleanup(){ GlobalLock();
>
> if( m_AllowList != NULL ){ delete m_AllowList; m_AllowList = NULL; } /// special headers if( m_ViaList != NULL ){ delete m_ViaList; m_ViaList = NULL; }
> .....
>
> I will continue searching. ¿Is there a way to configure PMEMORY_CHECK for release?
>
> Thanks for your help.
>
> Gustavo Curetti
>
>
> From: cur...@ho...To: jo...@op...; joe...@gm...; ope...@li...Subject: RE: [OpenSIPStack] FW: Memory Leak in Proxy and Full ModeDate: Mon, 23 Jun 2008 21:42:45 +0200
>
>
> Hi Joegen I think this is a memory leak because after all messages are sent the memory never go down.The code added in B2BUserAgent::Registrar::ProcessUpperRegKeepAlive() waits 5 seconds and sends the first Invite, then waits 5 seconds , sends the second Invite, waits 5 seconds more and finally sends 4998 Invites with a 40 ms delay between each one. void B2BUserAgent::Registrar::ProcessUpperRegKeepAlive(){ DWORD delay = 5000; DWORD numberOfPackets = 0; while( !m_UpperRegSync.Wait( delay ) ) { if(numberOfPackets < 5000) { 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: " + OString(numberOfPackets) + "@192.168.0.10\r\nMax-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 ) ); if (numberOfPackets > 1) delay = 40; numberOfPackets++; } }} The SIP timers B and H are set to 20 ms. #define SIP_TIMER_B 20 #define SIP_TIMER_H 20 After all messages are sent, I wait for a long time but the memory used doesn't decrease. I tried different settings: Case 1 5 sec -> 1 Invite -> 5 sec -> 1 Invite -> 5 sec -> 4998 Invite (40 ms) -> 5 min1)Mem Usage 8332k 8940k 9080k 9400k2)Mem Usage 8364k 8952k 8976k 9416k3)Mem Usage 8300k 8888k 8908k 9368k Case 2 5 sec -> 1 Invite -> 5 sec -> 1 Invite -> 5 sec -> 9998 Invite (40 ms) -> 5 min1)Mem Usage 8308k 8900k 9000k 9696k2)Mem Usage 8320k 8908k 8928k 9760k The leak is bigger when the amount of messages are higher.Case 3 5 sec -> 1 Invite -> 5 sec -> 1 Invite -> 5 sec -> 4998 Invite (30 ms) -> 5 min1)Mem Usage 8252k 8832k 8852k 9532k 2)Mem Usage 8256k 8844k 8872k 9536k Case 4 5 sec -> 1 Invite -> 5 sec -> 1 Invite -> 5 sec -> 4998 Invite (20 ms) -> 5 min1)Mem Usage 8276k 8864k 8884k 9728k 2)Mem Usage 8240k 8828k 8848k 9728k The leak is bigger when the delay is shorter. Case 5 5 sec -> 1 Invite -> 5 sec -> 1 Invite -> 5 sec -> 4998 Invite (80 ms) -> 5 min1)Mem Usage 8344k 8932k 8952k 9332k2)Mem Usage 8376k 8968k 8992k 9276k The only type of error in the log is : 102:42:36.691 ERR: [CID=0x0000] UDP Socket Read Error (Socket not connected) in case 1 appears 5219 times. I don't know if this error is linked with the leak. The log is attached. This behavior doesn't seem to be because of transactions waiting for timers. Maybe is because some type of conflict with shared resources or mutexs, but I really don't know. Any idea? If you need more info, please let me know. One more thing, I suggest to increase MAX_SIP_MESSAGE_LENGTH because i had problems with Invites with a lot of headers (record-routes, vias, etc) and long bodies. #define MAX_SIP_MESSAGE_LENGTH 2048 ----> #define MAX_SIP_MESSAGE_LENGTH 4000 Thanks for your help. Gustavo Curetti
>
>
>> Date: Sat, 21 Jun 2008 10:17:22 +0800> To: cur...@gm...> From: joe...@gm...> CC: joe...@gm...; ope...@li...> Subject: Re: [OpenSIPStack] FW: Memory Leak in Proxy and Full Mode> > Hi Gustavo,> > This time, I am not sure if what you are seeing in perfmon is actually a > leak or simply transactions waiting for timers to fire so it could > transition from Completed to Terminated state. I tried to fire the same > in my perfmon and the private mem leveled at around 80 in the graph. > Same place where yours leveled to a straight line. Can you share more > info why you think this is a leak?> > Joegen> > > Gustavo Curetti wrote:> > Hi Joegen,> > > > Another case of memory leak is attached. (no debugging mode).> > I send 5000 messages with 40 ms delay between each message.> > A jpg of the perfmon is attached too. (Private Bytes)> > > > Thanks for your help> >> > Gustavo Curetti> >> >> > ------------------------------------------------------------------------> >> > > Date: Fri, 20 Jun 2008 09:33:12 +0800> > > To: cur...@gm...; ope...@li...> > > From: joe...@gm...> > > CC: joe...@gm...> > > Subject: Re: [OpenSIPStack] FW: Memory Leak in Proxy and Full Mode> > >> > > Right! Patch is in CVS.> > >> > > Joegen> > >> > > Gustavo Curetti wrote:> > > > Hi Joegen,> > > >> > > > Thanks for your help. The other header which is missing in the > > SIPMessage::CleanUp() is the m_SessionExpires ("session-expires"). I > > will continue with my tests and let you know.> > > >> > > > Gustavo> > > >> > > >> > > >> > >> > >> > > > > -------------------------------------------------------------------------> > > 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> >> >> > ------------------------------------------------------------------------> > ¿Aburrido? Ingresá ya y divertite como nunca en MSN Juegos. MSN Juegos > > <http://juegos.ar.msn.com/>> > ------------------------------------------------------------------------> >> > 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
>>
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From: OpenSIPStack F. <ope...@op...> - 2008-06-27 02:01:06
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More information:
- I patched Logger.cxx (void LoggingIncrementingFileStream::InternalOpen()) to create the log file in the executable directory for Windows, since it's hard to know where the log file goes otherwise (I couldn't find it when running from Visual Studio)
- The log file shows the following occurring _after_ SoftPhoneManager::Terminate( PThread &, INT ) exits:
{quote}40:48:08.425 pcss.cxx(193) PWL: [CID=0x0000] PCSS Deleted PC sound system endpoint.
40:48:08.426 endpoint.cxx(214) PWL: [CID=0x0000] OpalEP pc endpoint destroyed.
40:48:08.434 DTL: [CID=0x0000] *** REMOVED TRANSACTION *** NICT|2dc...@pr...|z9hG4bK2dc1daefe5fb1810843ff7f0e31de16f|REGISTER
40:48:08.434 DBG: [CID=0x0000] GC: First Stale Object SIPTransaction
40:48:08.434 DTL: [CID=0x0000] *** REMOVED TRANSACTION *** NICT|2dc...@pr...|z9hG4bK4649e0efe5fb1810843ff7f0e31de16f|REGISTER
40:48:08.435 DBG: [CID=0x0000] GC: First Stale Object SIPTransaction
40:48:08.435 SIPTransactionStage.cxx(86) PWL: [CID=0x0000] Transaction Thread [2264] Terminated
40:48:08.436 SIPTransactionStage.cxx(122) PWL: [CID=0x0000] Transaction Cleaner Thread [2272] Terminated
40:48:08.436 SIPTransactionStage.cxx(122) PWL: [CID=0x0000] Transaction Cleaner Thread [2580] Terminated
40:48:08.438 SIPTimerManager.cxx(92) PWL: [CID=0x0000] *** DESTROYED *** SIPTimer Manager
40:48:08.438 endpoint.cxx(214) PWL: [CID=0x0000] OpalEP sip endpoint destroyed.
40:48:08.438 manager.cxx(401) PWL: [CID=0x0000] OpalMan Deleted manager.
40:48:08.439 DBG: [CID=0x11a8] TRANSACTION: (NICT) DESTROYED
40:48:08.439 DTL: [CID=0x11a8] NICT(146884423) *** DESTROYED *** - NICT|2dc...@pr...|z9hG4bK2dc1daefe5fb1810843ff7f0e31de16f|REGISTER
40:48:08.440 DBG: [CID=0x11a8] TRANSACTION: (NICT) DESTROYED
40:48:08.440 DTL: [CID=0x11a8] NICT(146884425) *** DESTROYED *** - NICT|2dc...@pr...|z9hG4bK4649e0efe5fb1810843ff7f0e31de16f|REGISTER
40:48:08.471 INF: [CID=0x0cb4] *** DESTROYED *** REGISTER Session REG...@pr...
40:48:08.472 DBG: [CID=0x0cb4] REGISTER: Session DESTROYED
40:48:08.484 DBG: [CID=0x11a8] TRANSACTION: (NICT) DESTROYED
40:48:08.484 DTL: [CID=0x11a8] NICT(146884424) *** DESTROYED *** - NICT|2dc...@pr...|z9hG4bK3d17dbefe5fb1810843ff7f0e31de16f|REGISTER
40:48:08.536 ECCIAPI.cxx(429) PWL: [CID=0x0000] Accept failed for HTTP: WIN32 error 10093
40:48:08.537 ECCIAPI.cxx(429) PWL: [CID=0x0000] Accept failed for HTTP: WIN32 error 10093
40:48:08.539 ECCIAPI.cxx(429) PWL: [CID=0x0000] Accept failed for HTTP: WIN32 error 10093
40:48:08.540 ECCIAPI.cxx(429) PWL: [CID=0x0000] Accept failed for HTTP: WIN32 error 10093
40:48:08.541 ECCIAPI.cxx(429) PWL: [CID=0x0000] Accept failed for HTTP: WIN32 error 10093
40:48:08.543 ECCIAPI.cxx(429) PWL: [CID=0x0000] Accept failed for HTTP: WIN32 error 10093
40:48:08.545 ECCIAPI.cxx(429) PWL: [CID=0x0000] Accept failed for HTTP: WIN32 error 10093
40:48:08.547 ECCIAPI.cxx(429) PWL: [CID=0x0000] Accept failed for HTTP: WIN32 error 10093
40:48:08.548 ECCIAPI.cxx(429) PWL: [CID=0x0000] Accept failed for HTTP: WIN32 error 10093
40:48:08.549 ECCIAPI.cxx(429) PWL: [CID=0x0000] Accept failed for HTTP: WIN32 error 10093
40:48:08.551 ECCIAPI.cxx(429) PWL: [CID=0x0000] Accept failed for HTTP: WIN32 error 10093
40:48:08.552 ECCIAPI.cxx(429) PWL: [CID=0x0000] Accept failed for HTTP: WIN32 error 10093
40:48:08.553 ECCIAPI.cxx(429) PWL: [CID=0x0000] Accept failed for HTTP: WIN32 error 10093
40:48:08.555 ECCIAPI.cxx(429) PWL: [CID=0x0000] Accept failed for HTTP: WIN32 error 10093
40:48:08.556 ECCIAPI.cxx(429) PWL: [CID=0x0000] Accept failed for HTTP: WIN32 error 10093
40:48:08.557 ECCIAPI.cxx(429) PWL: [CID=0x0000] Accept failed for HTTP: WIN32 error 10093
40:48:08.560 ECCIAPI.cxx(429) PWL: [CID=0x0000] Accept failed for HTTP: WIN32 error 10093
40:48:08.561 ECCIAPI.cxx(429) PWL: [CID=0x0000] Accept failed for HTTP: WIN32 error 10093
40:48:08.562 ECCIAPI.cxx(429) PWL: [CID=0x0000] Accept failed for HTTP: WIN32 error 10093
40:48:08.564 ECCIAPI.cxx(429) PWL: [CID=0x0000] Accept failed for HTTP: WIN32 error 10093
40:48:08.565 ECCIAPI.cxx(429) PWL: [CID=0x0000] Accept failed for HTTP: WIN32 error 10093
40:48:08.567 ECCIAPI.cxx(429) PWL: [CID=0x0000] Accept failed for HTTP: WIN32 error 10093
40:48:08.568 ECCIAPI.cxx(429) PWL: [CID=0x0000] Accept failed for HTTP: WIN32 error 10093
40:48:08.569 ECCIAPI.cxx(429) PWL: [CID=0x0000] Accept failed for HTTP: WIN32 error 10093
40:48:08.571 ECCIAPI.cxx(429) PWL: [CID=0x0000] Accept failed for HTTP: WIN32 error 10093
40:48:08.572 ECCIAPI.cxx(429) PWL: [CID=0x0000] Accept failed for HTTP: WIN32 error 10093
40:48:08.574 ECCIAPI.cxx(429) PWL: [CID=0x0000] Accept failed for HTTP: WIN32 error 10093
40:48:08.575 ECCIAPI.cxx(429) PWL: [CID=0x0000] Accept failed for HTTP: WIN32 error 10093
40:48:08.577 ECCIAPI.cxx(429) PWL: [CID=0x0000] Accept failed for HTTP: WIN32 error 10093
40:48:08.578 ECCIAPI.cxx(429) PWL: [CID=0x0000] Accept failed for HTTP: WIN32 error 10093
40:48:08.579 ECCIAPI.cxx(429) PWL: [CID=0x0000] Accept failed for HTTP: WIN32 error 10093
40:48:08.581 ECCIAPI.cxx(429) PWL: [CID=0x0000] Accept failed for HTTP: WIN32 error 10093
40:48:08.582 ECCIAPI.cxx(429) PWL: [CID=0x0000] Accept failed for HTTP: WIN32 error 10093
40:48:08.584 ECCIAPI.cxx(429) PWL: [CID=0x0000] Accept failed for HTTP: WIN32 error 10093
40:48:08.585 ECCIAPI.cxx(429) PWL: [CID=0x0000] Accept failed for HTTP: WIN32 error 10093
40:48:08.586 ECCIAPI.cxx(429) PWL: [CID=0x0000] Accept failed for HTTP: WIN32 error 10093
40:48:08.588 ECCIAPI.cxx(429) PWL: [CID=0x0000] Accept failed for HTTP: WIN32 error 10093
40:48:08.589 ECCIAPI.cxx(429) PWL: [CID=0x0000] Accept failed for HTTP: WIN32 error 10093
40:48:08.591 ECCIAPI.cxx(429) PWL: [CID=0x0000] Accept failed for HTTP: WIN32 error 10093
40:48:08.592 ECCIAPI.cxx(429) PWL: [CID=0x0000] Accept failed for HTTP: WIN32 error 10093
40:48:08.594 ECCIAPI.cxx(429) PWL: [CID=0x0000] Accept failed for HTTP: WIN32 error 10093
40:48:08.595 ECCIAPI.cxx(429) PWL: [CID=0x0000] Accept failed for HTTP: WIN32 error 10093
40:48:08.596 ECCIAPI.cxx(429) PWL: [CID=0x0000] Accept failed for HTTP: WIN32 error 10093
40:48:08.598 ECCIAPI.cxx(429) PWL: [CID=0x0000] Accept failed for HTTP: WIN32 error 10093
40:48:08.599 ECCIAPI.cxx(429) PWL: [CID=0x0000] Accept failed for HTTP: WIN32 error 10093
40:48:08.600 ECCIAPI.cxx(429) PWL: [CID=0x0000] Accept failed for HTTP: WIN32 error 10093
40:48:08.602 ECCIAPI.cxx(429) PWL: [CID=0x0000] Accept failed for HTTP: WIN32 error 10093{quote}
I don't know much about OpenSipStack, but this seems like a lot of cleanup activity going on after OpenSipStack has said it has terminated. In trying to track down the shutdown errors in OSSPhone (MFC) previously, I did get hits in ECCIAPI suggesting that a previously freed event handler was being triggered. There was also an issue with a timer triggering something. (Maybe they're related.) Is there an easy way to make sure this stuff is completed before SoftPhoneInterface:Event_Terminated() is triggered?
Finest regards,
Bill Root
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From: OpenSIPStack F. <ope...@op...> - 2008-06-26 16:09:35
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Joegen,
I have a Delphi version of OSSPhone largely implemented, but it gets an access violation on shutdown in ATLSTIP.DLL, like my test program.
I see that the debug version of OSSPhone (MFC) frequently hangs on shutdown when run from Visual Studio 2005, if it was registered with the SIP server. When I "Break All" from Visual Studio, I get the following error message:
{quote}"The process appears to be deadlocked (or is not running any user-mode code). All threads have been stopped."{quote}
This happened the last three times I tried it. I logged in (to the SIP server), and twice quit without logging out. The third time I logged out before quitting.
I'm running in a Virtual PC 2007 VM under Windows XP.
I'm using CVS source from 2008-06-25.
Do you know of any problems during shutdown with OSSPhone (MFC) or OpenSipStack?
Finest regards,
Bill
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