[asterisk-users] asterisk 11.7.0: Delayed audio

[Gerard Saraber]

I wanted to chime in on this one, I posted a similar problem a while
back under the heading "Delay before audio starts" on 2/26/2013

My solution to fix this problem was to adjust my dialplan by inserting
an Answer(); 
So I don't think it necessarily has something to do with the strictrtp
setting.

-Gerard

On Fri, 2014-01-10 at 15:01 -0600, Matthew Jordan wrote:
> On Fri, Jan 10, 2014 at 9:45 AM, gm1 <gm1 at curtissystemssoftware.com> wrote:
> > On connection to an incoming call via PSTN where
> > asterisk [11.7.0] is Dialing an internal extension
> > on answering the call there is about 6-7 seconds before
> > audio is heard on either side.
> >
> >
> > When looking at the CLI traces when I answer the incoming call that asterisk
> > extensions were dialing, I see immediately upon answering
> >>0xhexnumber -- Probation passed - setting RTP source address to
> >> 192.168.1.11:portnumber
> > then not until about 6 seconds later I see this
> >>0xhexnumber -- Probation passed - setting RTP source address to
> >> 192.168.1.11:diffportnumber
> > and immediately hear audio
> >
> > what appears to be an issue is that the RTP link(audio) setup is delayed.
> >
> >
> > Anyone have suggestions on how to fix this issue?
> >
> 
> If the RTP source address/port is changing, then Asterisk is receiving
> RTP packets from two different sources and is waiting for one of them
> to stabilize before it picks the actual source of the media stream.
> That's by design, as the "locking in" of the RTP source prevents a
> media injection attack.
> 
> You can tweak how Asterisk does this using two settings in rtp.conf:
> 
> ; Enable strict RTP protection. This will drop RTP packets that
> ; do not come from the source of the RTP stream. This option is
> ; enabled by default.
> ; strictrtp=yes
> 
> ; Number of packets containing consecutive sequence values needed
> ; to change the RTP source socket address. This option only comes
> ; into play while using strictrtp=yes. Consider changing this value
> ; if rtp packets are dropped from one or both ends after a call is
> ; connected. This option is set to 4 by default.
> ; probation=8
> 
> Matt
> 
> -- 
> Matthew Jordan
> Digium, Inc. | Engineering Manager
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
> Check us out at: http://digium.com & http://asterisk.org
> 

-- 
Gerard Saraber <gsaraber at rarcoa.com>
Rarcoa, Inc.