[asterisk-bugs] [JIRA] (ASTERISK-24127) rtpkeepalive sometimes sends comfort noise rtp packets unnecessarily

[Rusty Newton (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-24127?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=221444#comment-221444 ] 

Rusty Newton commented on ASTERISK-24127:
-----------------------------------------

So, using the milliwatt test and your configs I can confirm that the behavior your describe is accurate.

Though, I think it may be expected behavior. That is, I believe rtpkeepalive is expected to send the comfort noise packets regardless of what is happening with the RTP stream. If the far end device interprets them as noise and plays them then that is expected.

I'll check with some Asterisk developers to determine expected behavior and then we can clarify in the documentation if necessary.

> rtpkeepalive sometimes sends comfort noise rtp packets unnecessarily
> --------------------------------------------------------------------
>
>                 Key: ASTERISK-24127
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-24127
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Resources/res_rtp_asterisk
>    Affects Versions: 11.11.0
>            Reporter: alexr1
>            Assignee: Rusty Newton
>            Severity: Minor
>
> I was experiencing minor audio clipping/interruptions and when I did a packet capture I found an RTP Comfort Noise packet corresponding with each interruption (PT=comfort noise). An easy fix is to disable rtpkeepalive.
> In sip.conf, rtpkeepalive specifies the number of seconds of no rtp activity before sending a comfort noise rtp packet as a keep alive. In some cases, Asterisk sends one in the middle of rtp streams (It seems to send it to both parties simultaneously, too).
> directmedia=no, so all rtp traffic is being handled by both asterisk servers.
> Interruptions every 10 seconds:
> AST11 Playing MOH <alaw> AST11 <alaw> SIP Phone
> No Interruptions when transcoding takes place:
> AST11 Playing MOH <alaw> AST11 <ulaw> SIP Phone
> AST11 Playing MOH <ulaw> AST11 <alaw> SIP Phone
> Unfortunately I don't have time to help further isolate the issue on our production system - mainly posting this so that someone else can find it if they're looking for a solution!



--
This message was sent by Atlassian JIRA
(v6.2#6252)