[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

[zvision (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-24127?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=236636#comment-236636 ] 

zvision commented on ASTERISK-24127:
------------------------------------

In parallel, I found that SSRC mess is caused by P2P bridging not maintaining SSRC across the bridge,
so after 200 OK is sent, SSRC gets simply forwarded between RTP instances, and Asterisk generated
CN packets may hit a scenario, where no SSRC has been set on an RTP instance.

For example, if RTP is being sent after 200 OK it gets into P2P mode immediatelly, so its RTP instance
has no rxssrc and lastts fields set, that why CN packets may have its own SSRC and timestamp = 0 each.
Only the lastseq gets updated.

> 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
>            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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