How to get rid of stuttering Asterisk?

Question

Asterisk

Good afternoon.
Conducted ip telephony for the enterprise. We raised our server and distribute internal subscribers through it.
server - Ubuntu 20.04.2 LTS
Asterisk version - Asterisk 16.2.1 - pure asterisk

There are two problems with communication:
1 Echo to city numbers, but the provider stubbornly says that this is normal and depends on the equipment through which the channel is created.
2 - BASIC - stuttering on the channel. Not always, but very often the subscriber starts to stutter, it can happen both right from the first second and after a few minutes.
The problem is two-sided. both we and the subscriber stutter.

I checked for packet loss with the provider via mtr -
100 packets were lost in half an hour, the average ping is 20ms, the maximum 120ms

is the connection diagram:
configs :
sip conf :

[general]
deny=0.0.0.0/0.0.0.0
permit=192.168.0.44/255.255.255.0
permit=****/255.255.255.0
allowsubscribe=yes
callcounter=yes
subscribecontext=BLF

[authentication]

[20111]
type=friend
host=dynamic
secret=*****
nat=force_rport,comedia
canreinvite=no
context=internal
qualify=yes
call-limit=2
disallow=all
allow=alaw
deny=0.0.0.0/0.0.0.0
permit=192.168.0.0/255.255.255.0
allowsubscribe=yes
callcounter=yes
subscribecontext=BLF

extensions.conf

[general]
static=yes
writeprotect=no
[globals]
DBHost = localhost
DBuser = ********
DBpass = ********
DBname = asterisk
DBowners = owners

[default]

;Вешаем трубку
[handup-sip]
exten => _X!,1,HangUp()

; Busy Lamp Functionality
[BLF]
exten => _1XX,hint,SIP/20${EXTEN}

[internal]
;исходящие звонки с внутренних аппаратов попадают сюда

exten => _[123]XX,1,Set(fname=${STRFTIME(${EPOCH},,%Y-%m-%d-%H%M)}-${CALLERID(number)}-${EXTEN})
same => n,Set(AUDIOHOOK_INHERIT(MixMonitor)=yes)
same => n,Monitor(wav,/home/asterisk/records/inner/${fname},mb)
same => n,Dial(SIP/20${EXTEN},30)

[rset] ;входящие звонки с транка попадают сюда

exten => _201[123]X,1,Set(fname=${STRFTIME(${EPOCH},,%Y-%m-%d-%H%M)}-${CALLERID(number)}-${EXTEN})
same => n,Monitor(wav,/home/asterisk/records/in/${fname},mb)
same => n,Set(AUDIOHOOK_INHERIT(MixMonitor)=yes)
same => n,Playback(/home/asterisk/records/service/allRecord)
same => n,Dial(SIP/${EXTEN},30,rTt)

Answer 1

Actually, a fairly predictable problem with the delivery of packets through public channels with non-guaranteed delivery / delay.
A typical situation: a crowd of people with web browsing, asterisks, etc. The user chats on the phone and opens a page with a bunch of photos -> the browser starts dragging pictures into a bunch of streams. The provider's channel is not rubber and at some point overflows - packets queue up and wait ... For uploading pictures, this will slow down the display of the cat's photo a little at most, for voice - stuttering will occur.
There are methods to partially eliminate this problem:
- you can reserve part of the bandwidth for voice
- you can at least prioritize traffic up to and including your gateway
- use a separate channel / gateway / network for telephony
It is much more problematic to eliminate similar congestion on the provider's side. Well, guaranteed streams (the same E1) are expensive now ...