Hello PJSIP mailing list, We are a french startup and we have a problem with Asterisk/PJSIP. We didn't find any answer or solution in wiki so, we post our problem here. We have a business application that uses both conventional telephony and VoIP. We use the PJSIP library to make VoIP calls from mobile devices (Android & iOS). On server side we have Asterisk with PJSIP. Sometimes "Asterisk" process crash with "double free or corruption". This happens shortly after the INVITE transaction was finished (we hear about 0.5s of sound) and only if the call was started on Android device. We tried to reproduce the crash with other softphones (Zoiper, CSipSimple, Ekiga) and pjsua in CLI but it doesn't crash. Also it doesn't crash when iOS app is used. So, it seems that, the problem is with our Android implementation, but we don't know where to search for the solution. We tried workarounds from here: https://issues.asterisk.org/jira/browse/ASTERISK-25274 https://issues.asterisk.org/jira/browse/ASTERISK-25275 But nothing worked. This crash occur once in about 200 calls. After using Valgrind (valgrind.org) to analyze Asterisk memory, we restart Asterisk and crash is happening more often. Is there a link ? You will find backtrace and debug in attachments. We tried Asterisk versions: 13.14 and 14.2 PJSSIP versions: 2.5.5, 2.6 (We tried to change audio codec but nothing changed) Thanks a lot Adagio Team

Here are the files sorry... Le 10/03/2017 à 10:26, ➔ Dan ADAGIO a écrit : > Hello PJSIP mailing list, > > We are a french startup and we have a problem with Asterisk/PJSIP. > We didn't find any answer or solution in wiki so, we post our problem > here. > > We have a business application that uses both conventional telephony > and VoIP. > We use the PJSIP library to make VoIP calls from mobile devices > (Android & iOS). On server side we have Asterisk with PJSIP. > > Sometimes "Asterisk" process crash with "double free or corruption". > This happens shortly after the INVITE transaction was finished (we > hear about 0.5s of sound) and only if the call was started on Android > device. > > We tried to reproduce the crash with other softphones (Zoiper, > CSipSimple, Ekiga) and pjsua in CLI but it doesn't crash. Also it > doesn't crash when iOS app is used. So, it seems that, the problem is > with our Android implementation, but we don't know where to search for > the solution. > > We tried workarounds from here: > https://issues.asterisk.org/jira/browse/ASTERISK-25274 > https://issues.asterisk.org/jira/browse/ASTERISK-25275 > But nothing worked. > > This crash occur once in about 200 calls. > After using Valgrind (valgrind.org) to analyze Asterisk memory, we > restart Asterisk and crash is happening more often. Is there a link ? > > You will find backtrace and debug in attachments. > > We tried Asterisk versions: 13.14 and 14.2 > PJSSIP versions: 2.5.5, 2.6 > (We tried to change audio codec but nothing changed) > > Thanks a lot > Adagio Team > > _______________________________________________ > Visit our blog: http://blog.pjsip.org > > pjsip mailing list > pjsip@lists.pjsip.org > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org >

Hi Dan, I have the same issue with Asterisk. Can you add your files to the following asterisk issue: https://issues.asterisk.org/jira/browse/ASTERISK-26797 I haven't had the issue for a while now, I'm using the latest GIT release of the Asterisk 13 branch with bundled PJSIP. Kind regards, Ross ________________________________ From: pjsip <pjsip-bounces@lists.pjsip.org> on behalf of ➔ Dan ADAGIO <dan@studio-adagio.com> Sent: 10 March 2017 09:30 To: pjsip@lists.pjsip.org Subject: Re: [pjsip] Asterisk crashs with PJSIP Here are the files sorry... Le 10/03/2017 à 10:26, ➔ Dan ADAGIO a écrit : > Hello PJSIP mailing list, > > We are a french startup and we have a problem with Asterisk/PJSIP. > We didn't find any answer or solution in wiki so, we post our problem > here. > > We have a business application that uses both conventional telephony > and VoIP. > We use the PJSIP library to make VoIP calls from mobile devices > (Android & iOS). On server side we have Asterisk with PJSIP. > > Sometimes "Asterisk" process crash with "double free or corruption". > This happens shortly after the INVITE transaction was finished (we > hear about 0.5s of sound) and only if the call was started on Android > device. > > We tried to reproduce the crash with other softphones (Zoiper, > CSipSimple, Ekiga) and pjsua in CLI but it doesn't crash. Also it > doesn't crash when iOS app is used. So, it seems that, the problem is > with our Android implementation, but we don't know where to search for > the solution. > > We tried workarounds from here: > https://issues.asterisk.org/jira/browse/ASTERISK-25274 > https://issues.asterisk.org/jira/browse/ASTERISK-25275 [ASTERISK-25275] A11 SIGSEGV from pjnpath check_cached ...<https://issues.asterisk.org/jira/browse/ASTERISK-25275> issues.asterisk.org Asterisk; ASTERISK-25275; A11 SIGSEGV from pjnpath check_cached_response (ast_rtcp_read -> pj_stun_session_on_rx_pkt) > But nothing worked. > > This crash occur once in about 200 calls. > After using Valgrind (valgrind.org) to analyze Asterisk memory, we > restart Asterisk and crash is happening more often. Is there a link ? > > You will find backtrace and debug in attachments. > > We tried Asterisk versions: 13.14 and 14.2 > PJSSIP versions: 2.5.5, 2.6 > (We tried to change audio codec but nothing changed) > > Thanks a lot > Adagio Team > > _______________________________________________ > Visit our blog: http://blog.pjsip.org > > pjsip mailing list > pjsip@lists.pjsip.org > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org pjsip Info Page<http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org> lists.pjsip.org This is the mailing list to discuss pjsip, the open source sip stack. We also have answers to frequently asked questions. To see the collection of prior postings to ... >

------------------------------------------------------------------------ Hi Ross, Thank you very much for this fast answer. I'm happy to sew that we are not alone ! We will test and post our files I'll write here our results. Best regards Dan Le 10/03/2017 à 10:36, Ross Beer a écrit : > > Hi Dan, > > > I have the same issue with Asterisk. Can you add your files to the > following asterisk issue: > > > https://issues.asterisk.org/jira/browse/ASTERISK-26797 > > > I haven't had the issue for a while now, I'm using the latest GIT > release of the Asterisk 13 branch with bundled PJSIP. > > > Kind regards, > > > Ross > > > > ------------------------------------------------------------------------ > *From:* pjsip <pjsip-bounces@lists.pjsip.org> on behalf of ➔ Dan > ADAGIO <dan@studio-adagio.com> > *Sent:* 10 March 2017 09:30 > *To:* pjsip@lists.pjsip.org > *Subject:* Re: [pjsip] Asterisk crashs with PJSIP > Here are the files sorry... > > Le 10/03/2017 à 10:26, ➔ Dan ADAGIO a écrit : > > Hello PJSIP mailing list, > > > > We are a french startup and we have a problem with Asterisk/PJSIP. > > We didn't find any answer or solution in wiki so, we post our problem > > here. > > > > We have a business application that uses both conventional telephony > > and VoIP. > > We use the PJSIP library to make VoIP calls from mobile devices > > (Android & iOS). On server side we have Asterisk with PJSIP. > > > > Sometimes "Asterisk" process crash with "double free or corruption". > > This happens shortly after the INVITE transaction was finished (we > > hear about 0.5s of sound) and only if the call was started on Android > > device. > > > > We tried to reproduce the crash with other softphones (Zoiper, > > CSipSimple, Ekiga) and pjsua in CLI but it doesn't crash. Also it > > doesn't crash when iOS app is used. So, it seems that, the problem is > > with our Android implementation, but we don't know where to search for > > the solution. > > > > We tried workarounds from here: > > https://issues.asterisk.org/jira/browse/ASTERISK-25274 > <https://issues.asterisk.org/jira/browse/ASTERISK-25274> > > https://issues.asterisk.org/jira/browse/ASTERISK-25275 > <https://issues.asterisk.org/jira/browse/ASTERISK-25275> > [ASTERISK-25275] A11 SIGSEGV from pjnpath check_cached ... > <https://issues.asterisk.org/jira/browse/ASTERISK-25275> > issues.asterisk.org > Asterisk; ASTERISK-25275; A11 SIGSEGV from pjnpath > check_cached_response (ast_rtcp_read -> pj_stun_session_on_rx_pkt) > > > > > But nothing worked. > > > > This crash occur once in about 200 calls. > > After using Valgrind (valgrind.org) to analyze Asterisk memory, we > > restart Asterisk and crash is happening more often. Is there a link ? > > > > You will find backtrace and debug in attachments. > > > > We tried Asterisk versions: 13.14 and 14.2 > > PJSSIP versions: 2.5.5, 2.6 > > (We tried to change audio codec but nothing changed) > > > > Thanks a lot > > Adagio Team > > > > _______________________________________________ > > Visit our blog: http://blog.pjsip.org <http://blog.pjsip.org> > > > > pjsip mailing list > > pjsip@lists.pjsip.org > > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org > pjsip Info Page > <http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org> > lists.pjsip.org > This is the mailing list to discuss pjsip, the open source sip stack. > We also have answers to frequently asked questions. To see the > collection of prior postings to ... > > > > > > > > > _______________________________________________ > Visit our blog: http://blog.pjsip.org > > pjsip mailing list > pjsip@lists.pjsip.org > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org

Hi mailinglist, Hi Ross, We tested to use the latest GIT release like you did, and we launched stress tests this morning ... but it crashs again :( Anyway, thank you very much. We will continue to investigate and add more details in our issue https://issues.asterisk.org/jira/browse/ASTERISK-26853 If you or someone has any other idea ?!? Best regards Dan Le 10/03/2017 à 10:36, Ross Beer a écrit : > > Hi Dan, > > > I have the same issue with Asterisk. Can you add your files to the > following asterisk issue: > > > https://issues.asterisk.org/jira/browse/ASTERISK-26797 > > > I haven't had the issue for a while now, I'm using the latest GIT > release of the Asterisk 13 branch with bundled PJSIP. > > > Kind regards, > > > Ross > > > > ------------------------------------------------------------------------ > *From:* pjsip <pjsip-bounces@lists.pjsip.org> on behalf of ➔ Dan > ADAGIO <dan@studio-adagio.com> > *Sent:* 10 March 2017 09:30 > *To:* pjsip@lists.pjsip.org > *Subject:* Re: [pjsip] Asterisk crashs with PJSIP > Here are the files sorry... > > Le 10/03/2017 à 10:26, ➔ Dan ADAGIO a écrit : > > Hello PJSIP mailing list, > > > > We are a french startup and we have a problem with Asterisk/PJSIP. > > We didn't find any answer or solution in wiki so, we post our problem > > here. > > > > We have a business application that uses both conventional telephony > > and VoIP. > > We use the PJSIP library to make VoIP calls from mobile devices > > (Android & iOS). On server side we have Asterisk with PJSIP. > > > > Sometimes "Asterisk" process crash with "double free or corruption". > > This happens shortly after the INVITE transaction was finished (we > > hear about 0.5s of sound) and only if the call was started on Android > > device. > > > > We tried to reproduce the crash with other softphones (Zoiper, > > CSipSimple, Ekiga) and pjsua in CLI but it doesn't crash. Also it > > doesn't crash when iOS app is used. So, it seems that, the problem is > > with our Android implementation, but we don't know where to search for > > the solution. > > > > We tried workarounds from here: > > https://issues.asterisk.org/jira/browse/ASTERISK-25274 > <https://issues.asterisk.org/jira/browse/ASTERISK-25274> > > https://issues.asterisk.org/jira/browse/ASTERISK-25275 > <https://issues.asterisk.org/jira/browse/ASTERISK-25275> > [ASTERISK-25275] A11 SIGSEGV from pjnpath check_cached ... > <https://issues.asterisk.org/jira/browse/ASTERISK-25275> > issues.asterisk.org > Asterisk; ASTERISK-25275; A11 SIGSEGV from pjnpath > check_cached_response (ast_rtcp_read -> pj_stun_session_on_rx_pkt) > > > > > But nothing worked. > > > > This crash occur once in about 200 calls. > > After using Valgrind (valgrind.org) to analyze Asterisk memory, we > > restart Asterisk and crash is happening more often. Is there a link ? > > > > You will find backtrace and debug in attachments. > > > > We tried Asterisk versions: 13.14 and 14.2 > > PJSSIP versions: 2.5.5, 2.6 > > (We tried to change audio codec but nothing changed) > > > > Thanks a lot > > Adagio Team > > > > _______________________________________________ > > Visit our blog: http://blog.pjsip.org <http://blog.pjsip.org> > > > > pjsip mailing list > > pjsip@lists.pjsip.org > > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org > pjsip Info Page > <http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org> > lists.pjsip.org > This is the mailing list to discuss pjsip, the open source sip stack. > We also have answers to frequently asked questions. To see the > collection of prior postings to ... > > > > > > > > > _______________________________________________ > Visit our blog: http://blog.pjsip.org > > pjsip mailing list > pjsip@lists.pjsip.org > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org