Re: [asterisk-users] Possible bug in app_meetme.c
> What version are you running? 1.6.2.0-rc2 > Does that version support disabling talker optimization? Yes. > Have you tried disabling talker optimization? Yes. That's how I found the bug. I got no audio from the SIP phone into the conference, so I decided I'd try seeing if it did if the SIP phone specified no options. I did. So I turned them on one at time until I saw that it was the talker optimization that was causing the complete lack of audion. Then I looked at the only two uses of that flag alone and found what looked like a logic bug. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Possible bug in app_meetme.c
David Backeberg wrote: > From a quick glance at your patch, I would say that it probably tries > to address the audio quality problems I and others were experiencing. No, it's fixing a much more serious issue. As I sent to this list twice, when I have a conference between Dahdi ports and SIP phones, the people on the SIP phones were never heard. I had the 'o' option specified. The code, as written, says "for non-Dahdi (e.g., SIP phones), send to the channel if talking AND not optimizing talkers". It seems to me that it should send to the challel if talking OR not optimizing talkers. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Possible bug in app_meetme.c
On Sat, Oct 17, 2009 at 2:26 PM, Richard Kenner wrote: > Is this patch correct? The "&&" doesn't make logical sense to me. I think > it should be "||" and making this change fixes the problem I have with SIP > phones in MeetMe conferences. If it's correct, is there someplace more > formal that I should submit it to? So now I actually read through your patch. Please read my last post about getting up to speed with how the history of talker optimization with MeetMe() in 1.6 has gone. Having said that, I think you may be using talker optimization and that this was causing the same voice cut out problem I was having. What version are you running? Does that version support disabling talker optimization? Have you tried disabling talker optimization? I appreciate your work on this, but I fear your issues may have already been solved a rather long time ago. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Possible bug in app_meetme.c
On Sat, Oct 17, 2009 at 2:26 PM, Richard Kenner wrote:
> Is this patch correct? The "&&" doesn't make logical sense to me. I think
> it should be "||" and making this change fixes the problem I have with SIP
> phones in MeetMe conferences. If it's correct, is there someplace more
> formal that I should submit it to?
>
> *** app_meetme.c.old 2009-10-11 17:56:44.0 -0400
> --- app_meetme.c 2009-10-17 14:22:29.0 -0400
> ***
> *** 2901,2905
> to write out all the
> samples.
> */
> ! if (user->talking &&
> !(confflags & CONFFLAG_OPTIMIZETALKER)) {
> careful_write(fd,
> f->data.ptr, f->datalen, 0);
> }
> --- 2901,2905
> to write out all the
> samples.
> */
> ! if (user->talking ||
> !(confflags & CONFFLAG_OPTIMIZETALKER)) {
> careful_write(fd,
> f->data.ptr, f->datalen, 0);
> }
I have a limited acquaintance with the app_meetme.c code, having also
modified it for my own purposes in the past.
For some historical perspective, you should know that originally in
1.6, talker optimization was forced on, with no way to turn it off.
Many people reported that it sounded awful, and patches were released
to make optimization be off by default, with a flag to turn it on, the
way it was in 1.4
There are more details, especially as regards versions were various
patches from trunk were merged into the releases, but that's the very
quick version. If you want bug numbers, please check out
http://issues.asterisk.org and search for talker optimization and
meetme.
When I first tried to use MeetMe with 1.6, I had a lot of
voice-cut-out problems, which I was able to verify were caused by
over-aggressive talker optimization, resulting in me going to 1.6.1.*
series as that series at that time had the 'optimization off by
default' feature that I wanted.
>From a quick glance at your patch, I would say that it probably tries
to address the audio quality problems I and others were experiencing.
It's possible, and probably likely that somebody else already fixed
this. Please be sure to check through the issues, and make sure you
compare your work against app_meetme.c in trunk, as your work will NOT
make it into asterisk by posting it to asterisk-users. If you didn't
already know, you should know that SVN is often way ahead of the
1.6.0. series for many reasons. For the latest versions for each 1.6
series, you should check through:
http://svnview.digium.com/svn/asterisk/branches/
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Re: [asterisk-users] Possible bug in app_meetme.c
On Sat, Oct 17, 2009 at 2:26 PM, Richard Kenner wrote: > Is this patch correct? The "&&" doesn't make logical sense to me. I think > it should be "||" and making this change fixes the problem I have with SIP > phones in MeetMe conferences. If it's correct, is there someplace more > formal that I should submit it to? I don't know if it's correct since I don't know what the CONFFLAG_OPTIMIZETALKER flag does nor what your problem is. Giving more background info should help. If you want to submit a patch use issues.asterisk.org and read the guidelines before submitting the bug report. -- Moises Silva Software Developer Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3 Canada t. 1 905 474 1990 x 128 | e. m...@sangoma.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
