[asterisk-bugs] [JIRA] Commented: (ASTERISK-19995) Recording calls is strongly degraded when using RTP packetization of 60 ms (g729: 60)

[Rusty Newton (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-19995?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=195139#comment-195139 ] 

Rusty Newton commented on ASTERISK-19995:
-----------------------------------------

Suspended due to lack of activity. Please request a bug marshal in #asterisk-bugs on the IRC network irc.freenode.net to reopen the issue should you have the additional information requested.  Further information can be found at https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines



> Recording calls is strongly degraded when using RTP packetization of 60 ms (g729: 60)
> -------------------------------------------------------------------------------------
>
>                 Key: ASTERISK-19995
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-19995
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>    Affects Versions: 1.8.13.0
>         Environment: CentOS 5.6 or CentOS 5.8 on different hardware
>            Reporter: Artem Kalatsey
>            Assignee: Matt Jordan
>         Attachments: test_for_forum.7z
>
>
> When configuring RTP packetization, setting the timer to 60 ms (allow=g729: 60 in sip.conf). Heavily degraded call recording of a remote client, although the voice does not undergo any changes. When decreasing the timer recording quality become better, with a 20 ms difference in the voice channel is disappear. All works on real hardware, as the writer use mixmonitor. The problem is relevant for both version 1.6 and for 1.8
> Looked at different forums - any ideas to remedy is not found.

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