[asterisk-bugs] [JIRA] (ASTERISK-29997) Audio Related Issues with WebRTC over Kamailio integration with Asterisk

[Asterisk Team (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-29997?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=258585#comment-258585 ] 

Asterisk Team commented on ASTERISK-29997:
------------------------------------------

Thanks for creating a report! The issue has entered the triage process. That means the issue will wait in this status until a Bug Marshal has an opportunity to review the issue. Once the issue has been reviewed you will receive comments regarding the next steps towards resolution. Please note that log messages and other files should not be sent to the Sangoma Asterisk Team unless explicitly asked for. All files should be placed on this issue in a sanitized fashion as needed.

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> Audio Related Issues with WebRTC over Kamailio integration with Asterisk
> ------------------------------------------------------------------------
>
>                 Key: ASTERISK-29997
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-29997
>             Project: Asterisk
>          Issue Type: Information Request
>      Security Level: None
>          Components: . I did not set the category correctly.
>    Affects Versions: 18.11.0
>         Environment: Debian 10 Bullseye
> 3.5GB RAM
> 60GB HardDrive
> AzureCloud VM
>            Reporter: Kelvin Arinze E.
>            Severity: Critical
>              Labels: webrtc
>
> Hi,
> We setup our asterisk with Kamailio as our proxy server and sip firewall with Webrtc socket and RTP engine configured on it (the Kamailio Server). But we experience one way audio, no audio, hang up after 30s, when we try making calls between internal extensions and external calls as well.
> Is there any settings to be checked when integrating with Kamailio on asterisk? 



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