[asterisk-bugs] [JIRA] (ASTERISK-24127) chan_sip option rtpkeepalive results in comfort noise packets being sent despite flowing RTP - resulting in audible interruptions to audio

[Rusty Newton (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-24127?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=221458#comment-221458 ] 

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

[~jcolp] looked at the code and apparently it does appear that rtpkeepalive should only be sending out a keep alive if there is a gap in the RTP stream that exceeds the rtpkeepalive value.

In this case, it looks like rtpkeepalive is just sending out keep alive packets regardless of what is happening stream.

> chan_sip option rtpkeepalive results in comfort noise packets being sent despite flowing RTP - resulting in audible interruptions to audio
> ------------------------------------------------------------------------------------------------------------------------------------------
>
>                 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!



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