Toufic Khreish (Gmail)
2015-Mar-10 22:22 UTC
[asterisk-users] Asterisk 13.2.0 Video issues
Thank you, I needed a starting point to start my post. 1. Asterisk 12.8.1 (IAX2 voice issues) no video issues. Voice issues on IAX2 Trunks, All extensions are SIP. The IAX2 trunks on Asterisk 12.8.1 produces only one error out of : iax2 set debug trunk on [2015-03-10 16:28:42] WARNING[9614][C-0000000b]: chan_iax2.c:1793 compress_subclass: Can't compress subclass 2097217 On the box running asterisk 1.6.2.6 I receive the following warning: [2015-03-10 16:35:00] WARNING[24872]: translate.c:168 framein: no samples for alawtolin core show channels Channel Location State Application(Data) IAX2/Mypbx1-15288 (None) Up AppDial((Outgoing Line)) SIP/6000-0000000f (None) Up Dial(IAX2/Mypbx1/300,300,Tt) 2 active channels Trunks are between an asterisk 1.6.2.6 and asterisk 12.8.1 (IAX , Alaw and GSM codecs) Voice is not very clear and choppy If I try the same between an asterisk 13.2.0 and the asterisk 1.6.2.6 , voice is very clear. 2. Asterisk 13.2.0 Video issues (no IAX2 voice issues). Calls from Bria video sip phone (android or IOS) to Grandstream GXV3175 (asterisk engine stops/crashes) Call from Groundwire video sip (IOS since Android version does not H264 codec) to Grandstream GXV3175, Asterisk stops Calls between SIP Video softphones works well no issues. Calls from SIP video softphones (BRIA) to Grandstream GXV3275 works well. (Acrobitz Groundwire to GXV3275 picture on Grandstream is yellowish) Calls between GXV3275 and GXV3175 video streaming is very slow on the GXV3175 (this is not the case under Asterisk 12.8.1) Calls from GXV3175 to Bria (video is displayed on bria side only) There might be an issue on the Grandstream sip video phones as far as H264 is concerned however the case of streaming slowness is not there under Asterisk 12.8.1) I cannot find anything related to the moment where asterisk is breaking upon calling GXV3175 Best regards Khreish Toufic -----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Matthew Jordan Sent: Tuesday, March 10, 2015 1:20 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk 13.2.0 Video issues On Tue, Mar 10, 2015 at 4:15 AM, Toufic Khreish (Gmail) <toufic.khreish at gmail.com> wrote:> I recently compiled asterisk 13.2.0 on an RK3288 , I am suspecting > problems with the format H264, Asterisk 12.8.1 compiled on the same > hardware is behaving very well for the same format H264 > > Problem of asterisk 12.8.1 is IAX2 trunk bad voice quality. > > Could someone investigate the problem of Asterisk 13 with video > support on > H264 ? >There's no where near enough information in your e-mail to give someone an indication on where to start. What channels are involved? What are their configurations? What formats are negotiated on the channels? What symptoms do you see? What does the CLI show, both when active calls are running and for a 'core show channel' for the involved parties? -- Matthew Jordan Digium, Inc. | Director of Technology 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com & http://asterisk.org -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
On Tue, Mar 10, 2015 at 5:22 PM, Toufic Khreish (Gmail) <toufic.khreish at gmail.com> wrote:> Thank you, I needed a starting point to start my post. > > 1. Asterisk 12.8.1 (IAX2 voice issues) no video issues. > Voice issues on IAX2 Trunks, All extensions are SIP. > The IAX2 trunks on Asterisk 12.8.1 produces only one error out of : iax2 > set debug trunk on > [2015-03-10 16:28:42] WARNING[9614][C-0000000b]: chan_iax2.c:1793 > compress_subclass: Can't compress subclass 2097217 > > On the box running asterisk 1.6.2.6 I receive the following warning: > [2015-03-10 16:35:00] WARNING[24872]: translate.c:168 framein: no samples > for alawtolin > > > core show channels > Channel Location State Application(Data) > IAX2/Mypbx1-15288 (None) Up AppDial((Outgoing Line)) > SIP/6000-0000000f (None) Up > Dial(IAX2/Mypbx1/300,300,Tt) > 2 active channels > > Trunks are between an asterisk 1.6.2.6 and asterisk 12.8.1 (IAX , Alaw and > GSM codecs) > Voice is not very clear and choppy > > If I try the same between an asterisk 13.2.0 and the asterisk 1.6.2.6 , > voice is very clear.Both Asterisk 1.6.2 and Asterisk 12 no longer receive bug fixes, so I'm going to skip past this issue.> 2. Asterisk 13.2.0 Video issues (no IAX2 voice issues). > > Calls from Bria video sip phone (android or IOS) to Grandstream GXV3175 > (asterisk engine stops/crashes)Asterisk crashing is a bug. That's a bad thing. Please get a backtrace [1] and file an issue on the issue tracker [2]. A pcap of the message traffic would also be very helpful.> Call from Groundwire video sip (IOS since Android version does not H264 > codec) to Grandstream GXV3175, Asterisk stopsI'm going to assume "Asterisk stops" means it crashed as well. If you'd like to get a backtrace for that as well and attach it to the same issue, that would be helpful - it may be the same problem that you see with the Bria phone, or it may be something else.> Calls between SIP Video softphones works well no issues.Well, that's good. :-)> Calls from SIP video softphones (BRIA) to Grandstream GXV3275 works well. > (Acrobitz Groundwire to GXV3275 picture on Grandstream is yellowish) > Calls between GXV3275 and GXV3175 video streaming is very slow on the > GXV3175 (this is not the case under Asterisk 12.8.1) > Calls from GXV3175 to Bria (video is displayed on bria side only)Since there are some that work fine, and some that don't, the trick is going to be knowing: (1) How the SIP peers (or PJSIP endpoints) are configured (2) How the phones are negotiating media with Asterisk Both your SIP configuration as well as a DEBUG log - generated with trace logging, showing the negotiation [3] - will be needed to figure out what is occurring. [1] https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace [2] https://issues.asterisk.org/jira/ [3] https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information -- Matthew Jordan Digium, Inc. | Director of Technology 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com & http://asterisk.org
Toufic Khreish (Gmail)
2015-Mar-14 12:35 UTC
[asterisk-users] Asterisk 13.2.0 Video issues
I will rebuild my asterisk with the options enabled ONT_OPTIMIZE and BETTER_BACKTRACES Then I will create the traces and post them as per your recommendations. -----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Matthew Jordan Sent: Thursday, March 12, 2015 3:56 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk 13.2.0 Video issues On Tue, Mar 10, 2015 at 5:22 PM, Toufic Khreish (Gmail) <toufic.khreish at gmail.com> wrote:> Thank you, I needed a starting point to start my post. > > 1. Asterisk 12.8.1 (IAX2 voice issues) no video issues. > Voice issues on IAX2 Trunks, All extensions are SIP. > The IAX2 trunks on Asterisk 12.8.1 produces only one error out of : > iax2 set debug trunk on > [2015-03-10 16:28:42] WARNING[9614][C-0000000b]: chan_iax2.c:1793 > compress_subclass: Can't compress subclass 2097217 > > On the box running asterisk 1.6.2.6 I receive the following warning: > [2015-03-10 16:35:00] WARNING[24872]: translate.c:168 framein: no > samples for alawtolin > > > core show channels > Channel Location State Application(Data) > IAX2/Mypbx1-15288 (None) Up AppDial((Outgoing Line)) > SIP/6000-0000000f (None) Up > Dial(IAX2/Mypbx1/300,300,Tt) > 2 active channels > > Trunks are between an asterisk 1.6.2.6 and asterisk 12.8.1 (IAX , Alaw > and GSM codecs) Voice is not very clear and choppy > > If I try the same between an asterisk 13.2.0 and the asterisk 1.6.2.6 > , voice is very clear.Both Asterisk 1.6.2 and Asterisk 12 no longer receive bug fixes, so I'm going to skip past this issue.> 2. Asterisk 13.2.0 Video issues (no IAX2 voice issues). > > Calls from Bria video sip phone (android or IOS) to Grandstream > GXV3175 (asterisk engine stops/crashes)Asterisk crashing is a bug. That's a bad thing. Please get a backtrace [1] and file an issue on the issue tracker [2]. A pcap of the message traffic would also be very helpful.> Call from Groundwire video sip (IOS since Android version does not > H264 > codec) to Grandstream GXV3175, Asterisk stopsI'm going to assume "Asterisk stops" means it crashed as well. If you'd like to get a backtrace for that as well and attach it to the same issue, that would be helpful - it may be the same problem that you see with the Bria phone, or it may be something else.> Calls between SIP Video softphones works well no issues.Well, that's good. :-)> Calls from SIP video softphones (BRIA) to Grandstream GXV3275 works well. > (Acrobitz Groundwire to GXV3275 picture on Grandstream is yellowish) > Calls between GXV3275 and GXV3175 video streaming is very slow on the > GXV3175 (this is not the case under Asterisk 12.8.1) Calls from > GXV3175 to Bria (video is displayed on bria side only)Since there are some that work fine, and some that don't, the trick is going to be knowing: (1) How the SIP peers (or PJSIP endpoints) are configured (2) How the phones are negotiating media with Asterisk Both your SIP configuration as well as a DEBUG log - generated with trace logging, showing the negotiation [3] - will be needed to figure out what is occurring. [1] https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace [2] https://issues.asterisk.org/jira/ [3] https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information -- Matthew Jordan Digium, Inc. | Director of Technology 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com & http://asterisk.org -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Toufic Khreish (Gmail)
2015-Mar-16 23:12 UTC
[asterisk-users] Asterisk 13.2.0 Video issues
Hello Matthew, I have compiled Asterisk 13.2 with the following compiler Flags enabled: DON'T_OPTIMIZE DEBUG THREADS BETTER_BACKTRACES My asterisk is running with the asterisk_script: root 24048 39.4 2.4 128564 50640 pts/1 Sl 00:02 2:21 /usr/sbin/asterisk -f -vvvg -c core show locks ========================================================================= 13.2.0 === Currently Held Locks =========================================================================== <pending> <lock#> (<file>): <lock type> <line num> <function> <lock name> <lock addr> (times locked) ======================================================================== When my asterisk crashes there is no file called core. The results of gdb -se "asterisk" -ex "bt full" -ex "thread apply all bt" --batch -c core > /tmp/backtrace.txt /usr/src/asterisk-13.2.0/core: No such file or directory. No stack. What could be the problem ? Best regards Toufic -----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Matthew Jordan Sent: Thursday, March 12, 2015 3:56 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk 13.2.0 Video issues On Tue, Mar 10, 2015 at 5:22 PM, Toufic Khreish (Gmail) <toufic.khreish at gmail.com> wrote:> Thank you, I needed a starting point to start my post. > > 1. Asterisk 12.8.1 (IAX2 voice issues) no video issues. > Voice issues on IAX2 Trunks, All extensions are SIP. > The IAX2 trunks on Asterisk 12.8.1 produces only one error out of : > iax2 set debug trunk on > [2015-03-10 16:28:42] WARNING[9614][C-0000000b]: chan_iax2.c:1793 > compress_subclass: Can't compress subclass 2097217 > > On the box running asterisk 1.6.2.6 I receive the following warning: > [2015-03-10 16:35:00] WARNING[24872]: translate.c:168 framein: no > samples for alawtolin > > > core show channels > Channel Location State Application(Data) > IAX2/Mypbx1-15288 (None) Up AppDial((Outgoing Line)) > SIP/6000-0000000f (None) Up > Dial(IAX2/Mypbx1/300,300,Tt) > 2 active channels > > Trunks are between an asterisk 1.6.2.6 and asterisk 12.8.1 (IAX , Alaw > and GSM codecs) Voice is not very clear and choppy > > If I try the same between an asterisk 13.2.0 and the asterisk 1.6.2.6 > , voice is very clear.Both Asterisk 1.6.2 and Asterisk 12 no longer receive bug fixes, so I'm going to skip past this issue.> 2. Asterisk 13.2.0 Video issues (no IAX2 voice issues). > > Calls from Bria video sip phone (android or IOS) to Grandstream > GXV3175 (asterisk engine stops/crashes)Asterisk crashing is a bug. That's a bad thing. Please get a backtrace [1] and file an issue on the issue tracker [2]. A pcap of the message traffic would also be very helpful.> Call from Groundwire video sip (IOS since Android version does not > H264 > codec) to Grandstream GXV3175, Asterisk stopsI'm going to assume "Asterisk stops" means it crashed as well. If you'd like to get a backtrace for that as well and attach it to the same issue, that would be helpful - it may be the same problem that you see with the Bria phone, or it may be something else.> Calls between SIP Video softphones works well no issues.Well, that's good. :-)> Calls from SIP video softphones (BRIA) to Grandstream GXV3275 works well. > (Acrobitz Groundwire to GXV3275 picture on Grandstream is yellowish) > Calls between GXV3275 and GXV3175 video streaming is very slow on the > GXV3175 (this is not the case under Asterisk 12.8.1) Calls from > GXV3175 to Bria (video is displayed on bria side only)Since there are some that work fine, and some that don't, the trick is going to be knowing: (1) How the SIP peers (or PJSIP endpoints) are configured (2) How the phones are negotiating media with Asterisk Both your SIP configuration as well as a DEBUG log - generated with trace logging, showing the negotiation [3] - will be needed to figure out what is occurring. [1] https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace [2] https://issues.asterisk.org/jira/ [3] https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information -- Matthew Jordan Digium, Inc. | Director of Technology 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com & http://asterisk.org -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users