Re: [asterisk-users] Confbridge for 80 devices

[Sean Bright]

On 10/20/2022 5:35 PM, Jerry Geis wrote:
>
> ;dsp_drop_silence=yes  ; This option drops what Asterisk detects as
> silence from
>                        ; entering into the bridge.  Enabling this
> option will drastically
>                        ; improve performance and help remove the
> buildup of background
>                        ; noise from the conference. Highly recommended
> for large conferences
>                        ; due to its performance enhancements.

I would try adding this to all of your user profiles (type = user) and
see if that improves things.

Kind regards,
Sean



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Re: [asterisk-users] Confbridge for 80 devices

[Jerry Geis]

On Thu, Oct 20, 2022 at 6:15 PM Eric Wieling  wrote:

>
> https://wiki.asterisk.org/wiki/display/AST/Timing+Interfaces
>
>
Thanks - so based on this wiki - seems like "The only functionality that
requires internal timing is IAX2 trunking" - which I am not using .
Just ConfBridge... And getting crappy audio with about 80 devices and a 1 -
way conf.

other thoughts on what is happening ?


Jerry
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Re: [asterisk-users] Confbridge for 80 devices

[Eric Wieling]

https://wiki.asterisk.org/wiki/display/AST/Timing+Interfaces

On 10/20/22 17:35, Jerry Geis wrote:


[modules]
autoload = yes
noload = res_timing_pthread
noload = res_timing_timerfd

SO I "dont" want to load res_timing_anything ???

I have preload on res_timing_dahdi, then res_timing_pthread and not 
res_timing_timerfd at all.




confbridge.conf is below

[general]
; The general section of this config
; is not currently used, but reserved
; for future use.

;
; --- Default Information ---
; The default_user and default_bridge sections are applied
; automatically to all ConfBridge instances invoked without
; a user, or bridge argument.  No menu is applied by default.
;

; --- ConfBridge User Profile Options ---
[default_user]
type=user
;admin=yes     ; Sets if the user is an admin or not. Off by default.
;marked=yes    ; Sets if this is a marked user or not. Off by default.
;startmuted=yes; Sets if all users should start out muted. Off by default
;music_on_hold_when_empty=yes  ; Sets whether MOH should be played when only
                                ; one person is in the conference or 
when the
                                ; the user is waiting on a marked user 
to enter

                                ; the conference. Off by default.
;music_on_hold_class=default   ; The MOH class to use for this user.
;quiet=yes     ; When enabled enter/leave prompts and user intros are 
not played.
                ; There are some prompts, such as the prompt to enter a 
PIN number,
                ; that must be played regardless of what this option is 
set to.

                ; Off by default
;announce_user_count=yes  ; Sets if the number of users should be 
announced to the

                           ; caller.  Off by default.
;announce_user_count_all=yes ; Sets if the number of users should be 
announced to
                              ; all the other users in the conference 
when someone joins.
                              ; This option can be either set to 'yes' 
or a number.
                              ; When set to a number, the announcement 
will only occur
                              ; once the user count is above the 
specified number.
;announce_only_user=yes   ; Sets if the only user announcement should be 
played
                           ; when a channel enters a empty conference.  
On by default.
;wait_marked=yes   ; Sets if the user must wait for a marked user to 
enter before

                    ; joining the conference. Off by default.
;end_marked=yes ; This option will kick every user with this option set 
in their
                 ; user profile after the last Marked user exists the 
conference.


;dsp_drop_silence=yes  ; This option drops what Asterisk detects as 
silence from
                        ; entering into the bridge.  Enabling this 
option will drastically
                        ; improve performance and help remove the 
buildup of background
                        ; noise from the conference. Highly recommended 
for large conferences

                        ; due to its performance enhancements.

;dsp_talking_threshold=128  ; The time in milliseconds of sound above 
what the dsp has
                             ; established as base line silence for a 
user before a user
                             ; is considered to be talking.  This value 
affects several
                             ; operations and should not be changed 
unless the impact on

                             ; call quality is fully understood.
                             ;
                             ; What this value affects internally:
                             ;
                             ; 1. Audio is only mixed out of a user's 
incoming audio stream
                             ;    if talking is detected.  If this value 
is set too
                             ;    loose the user will hear themselves 
briefly each
                             ;    time they begin talking until the dsp 
has time to

                             ;    establish that they are in fact talking.
                             ; 2. When talk detection AMI events are 
enabled, this value
                             ;    determines when talking has begun 
which results in
                             ;    an AMI event to fire.  If this value 
is set too tight
                             ;    AMI events may be falsely triggered by 
variants in

                             ;    room noise.
                             ; 3. The drop_silence option depends on 
this value to determine
                             ;    when the user's audio should be mixed 
into the bridge
                             ;    after periods of silence.  If this 
value is too loose
                             ;    the beginning of a user's speech will 
get cut off as they

                             ;    transition from silence to talking.
                             ;
                             ; By default this value is 160 ms. V

Re: [asterisk-users] Confbridge for 80 devices

[Jerry Geis]

This is on the bare metal machine

 Recalculating Codec Translation (number of sample seconds: 1)

 Translation times between formats (in microseconds) for one second
of data
  Source Format (Rows) Destination Format (Columns)

   ulaw  alaw   gsm  g726 g726aal2 adpcm slin8 slin12 slin16 slin24
slin32 slin44 slin48 slin96 slin192 lpc10 speex8 speex16 speex32  ilbc
 g722 testlaw
 ulaw -  9150 15000 1500015000 15000  9000  17000  17000  17000
 17000  17000  17000  17000   17000 15000  15000   23000   23000 15000
17250   15000
 alaw  9150 - 15000 1500015000 15000  9000  17000  17000  17000
 17000  17000  17000  17000   17000 15000  15000   23000   23000 15000
17250   15000
  gsm 15000 15000 - 1500015000 15000  9000  17000  17000  17000
 17000  17000  17000  17000   17000 15000  15000   23000   23000 15000
17250   15000
 g726 15000 15000 15000 -15000 15000  9000  17000  17000  17000
 17000  17000  17000  17000   17000 15000  15000   23000   23000 15000
17250   15000
 g726aal2 15000 15000 15000 15000- 15000  9000  17000  17000  17000
 17000  17000  17000  17000   17000 15000  15000   23000   23000 15000
17250   15000
adpcm 15000 15000 15000 1500015000 -  9000  17000  17000  17000
 17000  17000  17000  17000   17000 15000  15000   23000   23000 15000
17250   15000
slin8  6000  6000  6000  6000 6000  6000 -   8000   8000   8000
  8000   8000   8000   80008000  6000   6000   14000   14000  6000
 82506000
   slin12 14500 14500 14500 1450014500 14500  8500  -   8000   8000
  8000   8000   8000   80008000 14500  14500   14000   14000 14500
14000   14500
   slin16 14500 14500 14500 1450014500 14500  8500   8500  -   8000
  8000   8000   8000   80008000 14500  145006000   14000 14500
 6000   14500
   slin24 14500 14500 14500 1450014500 14500  8500   8500   8500  -
  8000   8000   8000   80008000 14500  14500   14500   14000 14500
14500   14500
   slin32 14500 14500 14500 1450014500 14500  8500   8500   8500   8500
 -   8000   8000   80008000 14500  14500   145006000 14500
14500   14500
   slin44 14500 14500 14500 1450014500 14500  8500   8500   8500   8500
  8500  -   8000   80008000 14500  14500   14500   14500 14500
14500   14500
   slin48 14500 14500 14500 1450014500 14500  8500   8500   8500   8500
  8500   8500  -   80008000 14500  14500   14500   14500 14500
14500   14500
   slin96 14500 14500 14500 1450014500 14500  8500   8500   8500   8500
  8500   8500   8500  -8000 14500  14500   14500   14500 14500
14500   14500
  slin192 14500 14500 14500 1450014500 14500  8500   8500   8500   8500
  8500   8500   8500   8500   - 14500  14500   14500   14500 14500
14500   14500
lpc10 15000 15000 15000 1500015000 15000  9000  17000  17000  17000
 17000  17000  17000  17000   17000 -  15000   23000   23000 15000
17250   15000
   speex8 15000 15000 15000 1500015000 15000  9000  17000  17000  17000
 17000  17000  17000  17000   17000 15000  -   23000   23000 15000
17250   15000
  speex16 23500 23500 23500 2350023500 23500 17500  17500   9000  17000
 17000  17000  17000  17000   17000 23500  23500   -   23000 23500
15000   23500
  speex32 23500 23500 23500 2350023500 23500 17500  17500  17500  17500
  9000  17000  17000  17000   17000 23500  23500   23500   - 23500
23500   23500
 ilbc 15000 15000 15000 1500015000 15000  9000  17000  17000  17000
 17000  17000  17000  17000   17000 15000  15000   23000   23000 -
17250   15000
 g722 15600 15600 15600 1560015600 15600  9600  17500   9000  17000
 17000  17000  17000  17000   17000 15600  15600   15000   23000 15600
-   15600
  testlaw 15000 15000 15000 1500015000 15000  9000  17000  17000  17000
 17000  17000  17000  17000   17000 15000  15000   23000   23000 15000
17250   -
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Re: [asterisk-users] Confbridge for 80 devices

[Jerry Geis]

[modules]
autoload = yes
noload = res_timing_pthread
noload = res_timing_timerfd

SO I "dont" want to load res_timing_anything ???

I have preload on res_timing_dahdi, then res_timing_pthread and not
res_timing_timerfd at all.



confbridge.conf is below

[general]
; The general section of this config
; is not currently used, but reserved
; for future use.

;
; --- Default Information ---
; The default_user and default_bridge sections are applied
; automatically to all ConfBridge instances invoked without
; a user, or bridge argument.  No menu is applied by default.
;

; --- ConfBridge User Profile Options ---
[default_user]
type=user
;admin=yes ; Sets if the user is an admin or not. Off by default.
;marked=yes; Sets if this is a marked user or not. Off by default.
;startmuted=yes; Sets if all users should start out muted. Off by default
;music_on_hold_when_empty=yes  ; Sets whether MOH should be played when only
   ; one person is in the conference or when the
   ; the user is waiting on a marked user to
enter
   ; the conference. Off by default.
;music_on_hold_class=default   ; The MOH class to use for this user.
;quiet=yes ; When enabled enter/leave prompts and user intros are not
played.
   ; There are some prompts, such as the prompt to enter a PIN
number,
   ; that must be played regardless of what this option is set
to.
   ; Off by default
;announce_user_count=yes  ; Sets if the number of users should be announced
to the
  ; caller.  Off by default.
;announce_user_count_all=yes ; Sets if the number of users should be
announced to
 ; all the other users in the conference when
someone joins.
 ; This option can be either set to 'yes' or a
number.
 ; When set to a number, the announcement will
only occur
 ; once the user count is above the specified
number.
;announce_only_user=yes   ; Sets if the only user announcement should be
played
  ; when a channel enters a empty conference.  On
by default.
;wait_marked=yes   ; Sets if the user must wait for a marked user to enter
before
   ; joining the conference. Off by default.
;end_marked=yes ; This option will kick every user with this option set in
their
; user profile after the last Marked user exists the
conference.

;dsp_drop_silence=yes  ; This option drops what Asterisk detects as silence
from
   ; entering into the bridge.  Enabling this option
will drastically
   ; improve performance and help remove the buildup of
background
   ; noise from the conference. Highly recommended for
large conferences
   ; due to its performance enhancements.

;dsp_talking_threshold=128  ; The time in milliseconds of sound above what
the dsp has
; established as base line silence for a user
before a user
; is considered to be talking.  This value
affects several
; operations and should not be changed unless
the impact on
; call quality is fully understood.
;
; What this value affects internally:
;
; 1. Audio is only mixed out of a user's
incoming audio stream
;if talking is detected.  If this value is
set too
;loose the user will hear themselves
briefly each
;time they begin talking until the dsp has
time to
;establish that they are in fact talking.
; 2. When talk detection AMI events are
enabled, this value
;determines when talking has begun which
results in
;an AMI event to fire.  If this value is
set too tight
;AMI events may be falsely triggered by
variants in
;room noise.
; 3. The drop_silence option depends on this
value to determine
;when the user's audio should be mixed into
the bridge
;after periods of silence.  If this value
is too loose
;the beginning of a user's speech will get
cut off as they
;transition from silence to talking.
;
; By default this value is 160 ms. Valid values
are 1 through 2^31

;dsp_silence_threshold=2000 ; The time in milliseconds of sound falling
within the what
; the dsp has established as baseline silence

Re: [asterisk-users] Confbridge for 80 devices

[Stoyan Marinov]

Hi,

Dahdi timing is for dahdi hardware. See here: 
https://wiki.asterisk.org/wiki/display/AST/Timing+Interfaces 


You could check your asterisk modules using "module show" on asterisk cli.

Sounds like you might be doing transcoding, which might be the cause of your 
problems. Are you using the same codec for all calls? If so - which one? I'm 
not 100% sure about this, but asterisk might be using slin internally for the 
conf bridge, which means it might have to do some expensive transcoding on the 
receiving and then again on the sending end. You could check your transcoding 
times with "core show translation recalc".

Regards,
Stoyan

> On 21 Oct 2022, at 12:17 AM, Jerry Geis  wrote:
> 
> 
> What is the trick to get "preload => res_timing_dahdi" working ?
> 
> I have tried to add to both a CentOS 7 (metal box) and Ubuntu 20.04 (VMware 
> guest) system
> restart asterisk and neither print anything about res_timing_dahdi in the 
> /var/log/asterisk/messages file.
> 
> Both are having issues with around 80 Confbridge items. 
> 
> timing test on BOTH return the same...
> Attempting to test a timer with 50 ticks per second.
> Using the 'timerfd' timing module for this test.
> It has been 1000 milliseconds, and we got 50 timer ticks
> 
> I dont think either system is "correctly" using the dahdi timer.
> 
> Thoughts ?
> 
> jerry
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> 
> New to Asterisk? Start here:
>  https://wiki.asterisk.org/wiki/display/AST/Getting+Started
> 
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Re: [asterisk-users] Confbridge for 80 devices

[Sean Bright]

On 10/20/2022 5:17 PM, Jerry Geis wrote:

> What is the trick to get "preload => res_timing_dahdi" working ?

[modules]
autoload = yes
noload = res_timing_pthread
noload = res_timing_timerfd

However, it's unlikely to be a timing problem. Can you share your ConfBridge 
configuration?

Kind regards,
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Re: [asterisk-users] Confbridge for 80 devices

[Jerry Geis]

What is the trick to get "preload => res_timing_dahdi" working ?

I have tried to add to both a CentOS 7 (metal box) and Ubuntu 20.04 (VMware
guest) system
restart asterisk and neither print anything about res_timing_dahdi in the
/var/log/asterisk/messages file.

Both are having issues with around 80 Confbridge items.

timing test on BOTH return the same...
Attempting to test a timer with 50 ticks per second.
Using the 'timerfd' timing module for this test.
It has been 1000 milliseconds, and we got 50 timer ticks

I dont think either system is "correctly" using the dahdi timer.

Thoughts ?

jerry
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Re: [asterisk-users] Confbridge for 80 devices

[Jerry Geis]

On Thu, Oct 20, 2022 at 1:53 PM Jerry Geis  wrote:

> Bringing 80 devices into a conf bridge with CPU: AMD Phenom(tm) II X6
> 1045T Processor at 2.7G and audio is reported as staticy or not the best
> audio quality.
>
> Network is r8169 :02:00.0 eth0: RTL8168e/8111
> Link is 1G.
>
> Asterisk 18.14.0
>
> I would think this should be able to handle 80 calls (one way audio).
>
> How can I tell if asterisk is an able to handle this - or how can I find
> the bottle neck?
>
> Thanks
>
> Jerry
>



So I did the "timing test" got
 timing test
Attempting to test a timer with 50 ticks per second.
Using the 'timerfd' timing module for this test.


in modules I added
[modules]
autoload=yes
preload => res_timing_dahdi.so
preload => res_timing_pthread.so

Re-ran the timing test got the same thing.
Notice it says specifically timerfd - not timing_dahdi or timing_pthread.
System is CentOS 7.

lsmod | grep dahdi
lsmod | grep dahdi
dahdi_transcode14291  1 wctc4xxp
dahdi_voicebus 59241  1 wctdm24xxp
dahdi 228002  9
wctdm24xxp,wcaxx,dahdi_transcode,oct612x,dahdi_voicebus,wcb4xxp,wct4xxp,wcte43x,wcte13xp
crc_ccitt  12707  2 wctdm24xxp,dahdi

Is my timing OK - or not ?

Note /var/log/asterisk/messages says "nothing" about timing even though the
resourse were loaded.

Thoughts ?

Jerry
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[asterisk-users] Confbridge for 80 devices

[Jerry Geis]

Bringing 80 devices into a conf bridge with CPU: AMD Phenom(tm) II X6 1045T
Processor at 2.7G and audio is reported as staticy or not the best audio
quality.

Network is r8169 :02:00.0 eth0: RTL8168e/8111
Link is 1G.

Asterisk 18.14.0

I would think this should be able to handle 80 calls (one way audio).

How can I tell if asterisk is an able to handle this - or how can I find
the bottle neck?

Thanks

Jerry
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