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-------Original Message-------

From: dingdong

Date: 21/03/2012 02:57:02

To: FreeSWITCH Users Help

Subject: Re: [Freeswitch-users] NAT issues - Outbound Call drops after~30seconds

id like to know more bout wireshark and sip debugging,please include this on the conf call topics

---- On Tue, 20 Mar 2012 10:15:20 -0700 Michael Collins <msc@freeswitch.org> wrote ----

This is why I love Wireshark so much! Look at this purdee graph it makes:

See all those 200 OK's that your FS is sending to the Grandstream? Guess what your GS is sending in response to those: NADA! If you look at the BYE that FS sends to the GS you'll even see the reason:

SIP;cause=408;text=\"ACK Timeout\"

FS never gets an ACK back from the GS. So the question is: why? I'm unfamiliar with the GS so I'll have to defer to those with more experience than I. However, I think you'll find that tcpdumps and analyzing w/ Wireshark is extremely helpful. (Open the pcap, click "Telephony > VoIP calls" and then a new dialog opens. In this case it shows two calls - meaning two call legs. Click "Select All" then click "Flow" and you'll get the cool graph. Click around and see what other stuff does. :)

I'm thinking of doing a FreeSWITCH conference call presentation on the subject of collecting pcaps and doing Wireshark analysis. Let me know if you guys think that's a good presentation.

-MC

On Tue, Mar 20, 2012 at 10:00 AM, Brian Foster <bdfoster@endigotech.com> wrote:

Andrew,

root@homeserver:/usr/local/stund# ./client stunserver.org

STUN client version 0.97

Primary: Independent Mapping, Independent Filter, preserves ports, will hairpin

Return value is 0x000003

http://da1.endigovoip.com/dump.pcap

Kristian,

http://pastebin.freeswitch.org/18708

Michael,

I did replace the IP's for security purposes, but now I've realized that it's needed and it's not really that big of a deal. I'll end up changing the Flowroute creds after this is fixed up. The prior siptrace is exactly one call (two legs). I don't think it's a carrier issue, as I've tried calling a buddy's server direct sip with the same issues.

-BDF

On Tue, Mar 20, 2012 at 11:34 AM, Michael Collins <msc@freeswitch.org> wrote:

We have scores of machines behind NAT talking to Flowroute with no problems, so there's got to be something potentially non-obvious but easy that needs to be set/unset. I noticed in the SIP trace that there are several calls. It's hard to know what's what. I think your best bet is a pcap analyzed with Wireshark, as was mentioned elsewhere in this thread. I also noticed that you redacted IP addrs - you won't be able to do this with a pcap. If security is an issue then I'd say get the pcap and let us know here on the list, then those who can have a look will email you privately and you can send the pcap file to them.

-MC

On Mon, Mar 19, 2012 at 12:12 PM, Brian Foster <bdfoster@endigotech.com> wrote:

Alright, so I admit... I'm a little rusty when it comes to NAT, etc. I've only set up FS so far on machines with no NAT, so this is sort of a new experience for me.

I have a FreeSWITCH server located on the same local network as all of my phones here at the house. When I try to make a call to Flowroute, after about 30 seconds the call drops. It also does the exact same thing when I call a buddy's server directly via SIP.

Here's a siptrace of the call (I didn't think that the actual FS log would be much help):

http://pastebin.freeswitch.org/18697

...and here's a paste of 'sofia status':

http://pastebin.freeswitch.org/18698

...and just for good measure, here's a paste of vars.xml:

http://pastebin.freeswitch.org/18699

--
Brian D. Foster
Endigo Computer LLC
Email: bdfoster@endigotech.com
Phone: 317-800-7876
Indianapolis, Indiana, USA

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