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

[alexr1 (JIRA)]

     [ https://issues.asterisk.org/jira/browse/ASTERISK-24127?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]

alexr1 updated ASTERISK-24127:
------------------------------

    Description: 
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!

  was:
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!


> 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
>            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!



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