Audio To softphone drop in between of call

Asterisk Asterisk SIP
1

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.

2

Check where you are sending the rtp packets…

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

3

Chek is there any firewall blocks you.

Or you should Chek RTP Media IP.

4

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.

5

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
end?

Antony.

6

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

7

this is my pjsip config

PBX Core settings

Version: 18.16.0
Build Options: BUILD_NATIVE, OPTIONAL_API
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/asterisk.pid
    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

8

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