Summary: | ASTERISK-13563: jitterbuffer delays after a meetme finished | ||
Reporter: | Christopher Faust (cfaust) | Labels: | |
Date Opened: | 2009-02-11 10:47:22.000-0600 | Date Closed: | 2009-02-12 12:21:13.000-0600 |
Priority: | Major | Regression? | No |
Status: | Closed/Complete | Components: | Core/Jitterbuffer |
Versions: | Frequency of Occurrence | ||
Related Issues: | |||
Environment: | Attachments: | ||
Description: | The version of asterisk is 1.4. We need to explain why the jitter buffer is so high after being into a meetme. We don’t have this problem when we are not in a meetme IAX2/iax_internal_fi 192.168.90.101 iax_intern 00022/00120 00016/00020 00040ms 0133ms 0183ms gsm IAX2/iax_internal_fi 192.168.90.101 iax_intern 00028/00132 00032/00028 00040ms 0160ms 0200ms gsm IAX2/iax_internal_fi 192.168.90.101 iax_intern 00045/00133 00050/00051 00040ms 0207ms 0247ms gsm IAX2/iax_internal_fi 192.168.90.101 iax_intern 00055/00114 00008/00006 00000ms 0133ms 0185ms gsm IAX2/iax_internal_fi 192.168.90.101 iax_intern 00104/00097 00202/00205 00040ms 0207ms 0250ms gsm Voici l’extension qui dial ce meetme remote exten => _MEETME.,2,Dial(Local/j1${EXTEN}@${CONTEXT}/nj) exten => _j1MEETME.,1,Meetme(${EXTEN:8},Adqx) exten => _j1MEETME.,2,Hangup We need to explain why there is such a huge delay in the jitter buffer when in a meetme room ****** ADDITIONAL INFORMATION ****** version 1.4-r11047M on 64 bit machine | ||
Comments: | By: Leif Madsen (lmadsen) 2009-02-11 11:50:15.000-0600 You mention SVN revision 11047 -- which is so old that 1.4 wasn't even branched at that point. You also mention versions 1.4.17, and 1.4.23 in this bug report. Could you please clarify which version of Asterisk specifically you are using? In addition, a few changes to the jitterbuffer code have been put in recently, so I would encourage you to test this with the latest version from the 1.4 branch. Thanks! By: Christopher Faust (cfaust) 2009-02-11 14:23:58.000-0600 the version which has this issue is 1.4-r11047M By: Leif Madsen (lmadsen) 2009-02-11 15:32:56.000-0600 That revision certainly doesn't exist in the 1.4 branch. What does your 'show version' output? I suggest you upgrade to the latest from the 1.4 branch. By: Christopher Faust (cfaust) 2009-02-11 17:37:52.000-0600 the above is from the show version output I did not add built by root @ Lepton on a x86_64 running Linux on 2009-01-06 18:50:57 UTC By: Christopher Faust (cfaust) 2009-02-11 17:42:36.000-0600 basically i am going through 3 asterisk servers #1 in montreal #2 in fedricton #3 in another location #1 connects to #2 voice quality is fine then we do a meetme which brings in #3 when the meetme is finished and #1 is still connected to #3 that is where I see the delay causing extremely poor sound quality By: Leif Madsen (lmadsen) 2009-02-12 09:18:04.000-0600 We still have no idea what version you are actually running, because the version you supplied does not exist. Revision 11047 is too old to actually even be a 1.4 release. So either you're not running 1.4, or you're running something quite old, or some other combination of old or corrupt. I would encourage you to reproduce the issue on a more recent version of Asterisk. By: Clod Patry (junky) 2009-02-12 12:21:13.000-0600 blitzrage: forget about it. I know cfaust in real. I will take care of that. |