Audio To softphone drop in between of call

Hi We are running asterisk 18.16

Our Calls are having issue as the call progress in between the audio to softphone drop or at least feels liked dropped in recording in asterisk I can hear user saying hello hello, what should be the next step to validate what is happening.

Check where you are sending the rtp packets…

So I would suggest capture the call and analyse the resulting pcap file.

Chek is there any firewall blocks you.

Or you should Chek RTP Media IP.

Media RTP IP is same it’s not changing the RTP flow is fine for first X min ( X is not a defined value ) but after that in softphone <> asterisk leg it went completely silent but other sip provider <> other person sound is there.

Do you have a router in the path where you experience audio problems, which is
not in the path where the audio works fine, and does this router contain a “SIP
ALG” (Application Layer Gateway, sometimes ironically called a “SIP Helper”),
or is this router performing Network Address Translation?

If this is the case, I recommend you (a) turn off SIP ALG, (b) check the UDP
timeout settings on NAT, and (c) check whether NAT is set for “stateful” UDP
(it should be).

As a further diagnostic step, do you get different behaviour depending on
whether the call is initiated from the softphone, or initiated from the other


Thanks for the this but all was fine. just 2 day before we never faced this big issue now it’s there for almost every other call

this is my pjsip config

PBX Core settings

Version: 18.16.0
Maximum calls: Not set
Maximum open file handles: 50000
Root console verbosity: 0
Current console verbosity: 1
Debug level: 0
Trace level: 0
Dump core on crash: Yes
Core dump file: core
Maximum load average: 0.000000
Minimum free memory: 0 MB
Startup time: 14:32:05
Last reload time: 15:37:26
System: Linux/4.19.0-18-amd64 built by root on x86_64 2023-03-28 10:51:09 UTC
System name:
Entity ID: 00:50:56:03:19:ff
PBX UUID: f36f5221-cf4f-4086-8164-84a02d46be78
Default language: en
Language prefix: Enabled
User name and group: asterisk/asterisk
Running directory: /
Executable includes: Disabled
Transcode via SLIN: Enabled
Transmit silence during rec: Disabled
Generic PLC: Enabled
Generic PLC on equal codecs: Disabled
Hide Msg Chan AMI events: Disabled
Min DTMF duration:: 80
Cache media frames: Enabled
RTP use dynamic payloads: 1
RTP dynamic payload types: 35-63,96-127

  • Subsystems

    Manager (AMI): Enabled
    Web Manager (AMI/HTTP): Enabled
    Call data records: Enabled
    Realtime Architecture (ARA): Disabled

  • Directories

    Configuration file: /etc/asterisk/asterisk.conf
    Configuration directory: /etc/asterisk
    Module directory: /usr/lib/asterisk/modules
    Spool directory: /var/spool/asterisk
    Log directory: /var/log/asterisk
    Run/Sockets directory: /var/run/asterisk
    PID file: /var/run/asterisk/
    VarLib directory: /var/lib/asterisk
    Data directory: /var/lib/asterisk
    ASTDB: /var/lib/asterisk/astdb
    IAX2 Keys directory: /var/lib/asterisk/keys
    AGI Scripts directory: /var/lib/asterisk/agi-bin

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