There is audibility in one direction. How to overcome?

Question

Asterisk

Good night all. There is Asterisk 13 + Addpac GS-1002. They are on the local network. SIP ALG is disabled on Mikrotik
Here are their configs:
sip.conf

[general]
trustrpid=yes
tcpenable=yes
useragent=D-link ; типа реальное железо. чтобы меньше привлекать внимание хацкеров которые ищут астериски
externip=внешний ip
localnet=192.168.5.0/24
qualify=yes               ; проверка доступности абонента - 2s - если больше, что считаем что недоступен
prematuremedia = no
progressinband = never
srvlookup=no
canreinvite=no              ; разрешает (yes) или запрещает (no) установку прямого соединения(минуя Asterisk).
directmedia=no              ; гнать трафик напрямую
allowguest = no             ; запрет регистрации "левых" аккунтов
transfer=yes                ; запрет трансфера вызовов глобально, включать вручную для нужных пиров
allowsubscribe=no           ; отказ от использования voicemail и соответствующего спама в консоли
alwaysauthreject=yes        ; на REGISTER Asterisk станет отвечать «401 Unathorized»
jbenable = yes              ; Enables the use of a jitterbuffer on the receiving side of a SIP
jbforce = no                ; Forces the use of a jitterbuffer on the receive side of a SIP
jbmaxsize=300
jbimpl = adaptive           ; Jitterbuffer implementation, used on the receiving side of a SIP
jbresyncthreshold=100       ;adaptive jbimpl had troubles without thresh... Asterisk 1.6.2.9.
;jblog = yes                ; Enables jitterbuffer frame logging. Defaults to "no".
context=default             ; всех левых в дефротный контекст на отбой
relaxdtmf=yes
dtmfmode=auto
disallow=all
;allow=g729
;allow=g723
allow=alaw
allow=ulaw
allow=gsm
bindport=5182



[addpac_channels](!)            ; шаблон дублирующихся настроек для каналов шлюза
host=dynamic
;deny=0.0.0.0/0
permit=192.168.5.110
fromdomain=192.168.5.110
type=friend
context=from_trunk                ; входящие с SIP попадают в этот контекст в extensions.conf
qualify=yes
nat=no
canreinvite=no
insecure=port,invite            ; игнорировать порт и инвайт
disallow=all
allow=alaw
allow=ulaw
allow=gsm
maxcallbitrate=64
dtmfmode=rfc2833
port=5182


[79640095533](addpac_channels)
defaultuser=79640095533
secret=*******
call-limit=2
callerid=79640095533
relaxdtmf=yes

[79298835533](addpac_channels)
defaultuser=79298835533
secret=******
call-limit=1
callerid=79298835533
relaxdtmf=yes

[my_sip_user](!)
type=friend                   ; входящие и исходящие
Call-limit=2                  ; лимит количества одновременных звонков
host=dynamic                  ; обязательная регистрация
nat=force_rport,comedia       ; используется ли натирование адресов?
canreinvite=no                ; разрешает (yes) или запрещает (no) установку прямого соединения между участниками (минуя Asterisk).
directmedia=no                ; гнать трафик напрямую
dtmfmode=auto
disallow=all                  ; запретить все кодеки
;allow=alaw
;allow=ulaw
;allow=gsm                     ; разрешить нужные
;allow=g729
;allow=g723
port=5182
insecure=invite,port
context=_sip                  ; Контекст плана набора в extensions.conf, в который изначально попадают звонки с GSM-линий
transfer=yes
;transport=tcp                             ; В него включен основной контекст _sip для всего SIP-направления.

When sip show both peers register
Addpac config

!
! APOS(tm) configuration saved from vty
!  2016/11/06 19:41:52
!
version 8.51.011
!
hostname GS1002
!
username *****
!
!
script ntpdate default
 server ip time.nist.gov
 server ip time.windows.com
!
interface Loopback0
 ip address 127.0.0.1 255.0.0.0
!
interface FastEthernet0/0
 ip address 192.168.5.110 255.255.255.0
 speed auto
 no qos-control
!
interface FastEthernet0/1
 ip address 192.168.10.1 255.255.255.0
 speed auto
 no qos-control
!
ip route 0.0.0.0 0.0.0.0 192.168.5.254
!
!
!
!
http server
!
logging command
logging event 4-warning
logging on
!
!
!
!
! VoIP configuration.
!
!
! Voice service voip configuration.
!
voice service voip
 protocol sip
 dtmf-relay rfc-2833
 fax protocol t38 redundancy 0
 fax rate 9600
 h323 call start fast
 h323 call tunnel enable
 no call-barring unconfigured-ip-address
 no voip-inbound-call-barring enable
!
!
! Voice port configuration.
!
! GSM
voice-port 0/0
 connection plar 79298835533
 ring detect-timeout 70
 dial-tone-generate
 caller-id enable
 caller-id type etsi
!
!
! GSM
voice-port 0/1
 connection plar 79640095533
 ring detect-timeout 70
 dial-tone-generate
 caller-id enable
 caller-id type etsi
!
!
! FXO
voice-port 0/2
 no caller-id enable
!
!
! FXO
voice-port 0/3
 no caller-id enable
!
!
!
!
! service port group configuration.
!
!
!
! Pots peer configuration.
!
dial-peer voice 0 pots
 destination-pattern 00T
 port 0/0
 call-waiting
 user-name 79298835533
 user-password ****
 translate-outgoing called-number 0
 diversion 1
!
dial-peer voice 1 pots
 destination-pattern 01T
 port 0/1
 call-waiting
 user-name 79640095533
 user-password ***
 translate-outgoing called-number 1
 preference 2
 diversion 2
!
!
!
! Voip peer configuration.
!
dial-peer voice 2000 voip
 destination-pattern T
 session target sip-server
 session protocol sip
 voice-class codec 1
 no vad
 dtmf-relay rtp-2833
 description asterisk
!
!
!
dial-peer call-hold h
dial-peer call-transfer h
!
!
!
gatekeeper
!
!
! Gateway configuration.
!
gateway
 h323-id voip.192.168.5.100
 no ignore-msg-from-other-gk
!
!
! Codec classes configuration.
!
voice class codec 0
 codec preference 1 g711alaw
 codec preference 2 g711ulaw
 codec preference 3 g729
!
voice class codec 1
 codec preference 1 g711alaw
 codec preference 2 g711ulaw
 codec preference 3 g729
!
!
!
! Translation Rule configuration.
!
translation-rule 0
 rule 0      007T                     8T
!
translation-rule 1
 rule 0      017T                     8T
!
!
!
! SIP UA configuration.
!
sip-ua
 user-register
 sip-username *******
 sip-password *******
 sip-server 192.168.5.100 5182 1
 timeout treg 400
 called-party-number to-field
 remote-party-id
 session-refresh update
 register e164
!
!
! Tones
!
!
! SMS delivery configuration
!
sms-delivery
!
!
!
!
voip-interface ip FastEthernet0/0
!
line console
!
line vty
!
mobile dev-restart-by-unreg 180
mobile failed-call-retry 0
mobile ussd inter-frame-gap 100
mobile ussd balance-interval 120
mobile ussd retry-count 2
mobile ussd retry-interval 5
mobile ussd response-protection-time 5
mobile dev-restart-by-unknown-error
mobile cell-monitor 30
!
mobile 0/0
 gsm sms-language utf8
!
mobile 0/1
 gsm sms-language utf8
!

Very often there are situations when you are talking with a subscriber and suddenly there is audibility in one direction. Can't hear the operator. I took a dump and using Wireshark I look at it and this is the situation on the problem call

RTP dump

As I understand it, the jitter is to blame or am I wrong? What would you advise to do in such a situation?

Answer 1

Link to a dump of a broken call

Answer 2

So to speak, it's pure neighing:
localnet=192.168.5.0/24
Try to decompose the mask into
255.255.255.0 through NAT your traffic does not go.
Doesn't he walk? No VLAN? And then you never know ...
Check the addresses of the gateways on all devices. Disable all sorts of stun ice and others in the operator's phone / softphone