[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:comment-tabpanel&focusedCommentId=221216#comment-221216 ] 

alexr1 commented on ASTERISK-24127:
-----------------------------------

Both servers have:

{noformat}
; RTP Configuration
[general]
;
; RTP start and RTP end configure start and end addresses
;
rtpstart=10000
rtpend=20000
{noformat}

Both using realtime on separate tables, and the peer definitions are:

First server:

{noformat}
[general]
context = from-sip-external ; Send unknown SIP callers to this context
realm = XXX.XXX.XXX
port = 5060           ; Port to bind to (SIP is 5060)
bindaddr = XXX.XXX.XXX.XXX    ; Address to bind to (all addresses on machine)
srvlookup=no
dtmfmode=rfc2833
relaxdtmf=no
rfc2833compensate=yes
disallow=all
allow=g722
allow=g729
allow=alaw
allow=ulaw
callerid=Anonymous
rtpkeepalive=10
rtptimeout=60
progressinband=yes
rtsavesysname=yes
allowsubscribe=no
allowtransfer=no
tcpenable=yes
tcpbindaddr=XXX.XXX.XXX.XXX

t38pt_udptl=yes,redundancy,maxdatagram=400
t38pt_rtp=no
t38pt_tcp=no

rtcachefriends=yes
rtupdate=yes

[XXXXPBX]
type=friend
qualify=yes
nat=no
host=XXX.XXX.XXX.XXX
dtmfmode=rfc2833
context=from-internal-pbx
directmedia=no
canreinvite=no

{noformat}



Second Server:

{noformat}
[general]
context = from-sip-external ; Send unknown SIP callers to this context
realm = XXX.XXX.XXX
port = 5060           ; Port to bind to (SIP is 5060)
bindaddr = XXX.XXX.XXX.XXX    ; Address to bind to (all addresses on machine)
srvlookup=no
dtmfmode=rfc2833
relaxdtmf=no
rfc2833compensate=yes
disallow=all
allow=g729
allow=ulaw
allow=alaw
callerid=Anonymous
rtpkeepalive=10
rtptimeout=60
progressinband=yes
sendrpid=yes
videosupport=yes

;FOR BLF
rtcachefriends=yes
rtupdate=yes

notifyringing=yes
allowsubscribe=yes
callcounter=yes

[XXXXtel]
type=friend
insecure=very
host=XXX.XXX.XXX.XXX
fromuser=XXXXPBX
fromdomain=XXX.XXX.XXX.XXX
dtmfmode=rfc2833
disallow=all
context=pbx-incoming
canreinvite=no
directmedia=no
allow=ulaw
allow=alaw
allow=g729
allow=g722

{noformat}

Hold music was wave, (although a wav16 version was also present and it seems Asterisk sometimes plays the 16hkz file at 8khz but that's another issue).

The only change I made to work around the issue was to comment out the rtpkeepalive.

> 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!



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