Meetme conference quality

We are using meetme in asterisk and

Quality in conference call is bad. Sort of hard to discribe
Its muffled and scratchy , calling same numbers directly and doing daisy chain of call, quality is fine (one person adds another)

Below are some system details.

asterisk -rx ‘dahdi show version’
DAHDI Version: 2.11.1 Echo Canceller:

asterisk -rx ‘core show version’
Asterisk 11.20.0 built by root @ scas3 on a x86_64 running Linux on 2017-11-30 19:57:24 UTC

cat /proc/version
Linux version 3.16.0-4-amd64 ( (gcc version 4.8.4 (Debian 4.8.4-1) ) #1 SMP Debian 3.16.39-1 (2016-12-30)

We have server with 32 cores, dedicated for doing voip with 64GB of RAM. Server clocks around 300 calls on average and there are no quality issues.

  • What is causing this quality issue ?
  • How do we track this and fix it ?
  • Any more info I can provide here ?

You haven’t even told us the channel technology, or codecs!

Hi David,

Sorry i was not sure what else i could provide.

Here is Codecs. there were 5 participants I have checked 200 OKs and all used G711A

G711A - Part A
G711A - Part B
G711A - Part C
G711A - Part D
G711A - Part E

I dont know how to find “channel technology” Can you let me know how can i see that and i will respond ?

Please provide the characters up to the first “/” from a channel name. If they are DAHDI, also please indicate whether analogue, or digital.

Hi David,

Thanks for your quick response. Here is some info.

Its Digital - These calls are incoming from either other SIP phones or Mobile routing to our gateway via upstream.

Please let me know if i am still missing something ?

There were some test calls done around below time. All of them are showing DHADI

[May 3 13:49:22] VERBOSE[6201][C-000077b8] pbx.c: – Executing [s@macro-stdexten:503] MeetMe(“SIP/scns2-00013b1d”, “61289608612,ciMdsr”) in new stack
[May 3 13:49:24] VERBOSE[46378] file.c: – <DAHDI/pseudo-780770900> Playing ‘/var/spool/asterisk/meetme/meetme-username-61289608612-4.slin’ (language ‘en’)

[May 3 13:44:20] VERBOSE[46378] file.c: – <DAHDI/pseudo-780770900> Playing ‘/var/spool/asterisk/meetme/meetme-username-61289608612-3.slin’ (language ‘en’)
[May 3 13:46:17] VERBOSE[46499][C-000073fe] pbx.c: == Spawn extension (default, 61289608612, 150) exited non-zero on ‘SIP/scns1-0001323d’
[May 3 13:46:17] VERBOSE[46378] file.c: – <DAHDI/pseudo-780770900> Playing ‘/var/spool/asterisk/meetme/meetme-username-61289608612-2.slin’ (language ‘en’)
[May 3 13:46:40] VERBOSE[46305][C-000073f0] pbx.c: == Spawn extension (default, 61289608612, 150) exited non-zero on ‘SIP/scns1-00013218’
[May 3 13:46:40] VERBOSE[46378] file.c: – <DAHDI/pseudo-780770900> Playing ‘/var/spool/asterisk/meetme/meetme-username-61289608612-1.slin’ (language ‘en’)
[May 3 13:48:46] VERBOSE[5825][C-00007783] pbx.c: – Executing [61289608612@default:1] NoOp(“SIP/scns2-00013ab3”, “”) in new stack

It appears it is not DAHDI, so the Analogue/Digital question doesn’t apply. However there are two SIP channel drivers and you haven’t said which. The log shows it is the obsolescent chan_sip.l

Why are you using meetme. That is also on its way out.

Do you have internal timing enabled?