[asterisk-users] inbound Audio problems probably not NAT related?

[Steve Davies]

On Jan 15, 2008 8:12 PM, John Millican
<jmillican at sentinelcommunications.com> wrote:
> Hello all,
> Was hoping to get a sanity check along with a question.  Below is the
> output from top run with normal defaults, except to show both CPU's, on
> a SuSE 10.2 box with Asterisk v1.4.15.
>
[snip massive hardware spec]

We have locations which run 120 simultaneous PRI calls on less than
half of the specification you gave :) I don't think that 6-12 calls
will overload it!

> On inbound calls I lose the incoming audio after a couple minutes,
> outbound audio is always good, then after a while inbound audio
> magically starts up again. this happens on maybe 10% of calls at its
> worst. I have looked at the possibility of NAT issues and do not believe
> that to be the case.

First place to look here is for a duplicate IP address on the network.
An arp cache timeout (usually about 10 minutes) can caused the voice
packets to start going to the wrong place temporarily.

> I have noticed that the memory usage climbs steadily but I believe that
> is the kernel as top show no process with more than 0.4% memory usage.
> Although when I rebooted (yes, an act of desperation) over the weekend
> the amount of calls with this problem dropped dramatically along with
> total memory usage which is slowly climbing again.  Started at about
> 1gig on Saturday morning and is now at the 2.6gig shown above in top.

LOL. If I am reading it correctly, 2.4Gb of that is cache and buffers,
and therefore "does not count". This is an example of a Linux kernel
using memory effectively to improve performance.

> This box typically does around 35,000 minutes of calls each month with a
> couple "busy" periods each day during weekdays.  Normally no more than
> 10 to 12 calls at one time.
>
> provider-->T1 to Cisco router-->Asterisk-->phones

If you are using a Cisco switch, check that all of the silly Cisco
trunking modes are disabled on the port(s) used by Asterisk and the
phones, and ensure that fast-start is enabled for those ports too.
IMHO Cisco nearly always set the defaults for these features back to
front!

Check the keepalive period on the firewall for NAT sessions, and
perhaps disabling silence supression if it is enabled will help to
keep the NAT connection open.

> The router is doing NAT and routing all traffic from a specific IP to
> the asterisk box and dropping everything from any other IP.
> canreinvite is set to no on the sip trunk and all the phones.
>
> One thing that may be related is that when I ssh into this box it takes
> a full minute respond after the pass phrase is typed in.  Could this be
> related or am I just grasping at straws?

This usually means that you have not configured DNS correctly either
at the server end, or the DNS records for the client end are somehow
"lacking"

>
> Any Ideas?

That is a quick random braindump. Perhaps some of it will be useful :)

Steve