[Freeswitch-users] SIP Trace Question / NAT one way audio problem.

[Ken Rice]

Re-registering every 23 seconds isnt that big of a deal if the phones are
behind nat, sometimes this is needed due to the way NAT happens on UDP with
various routers. There is no way to track the start/end of the session w/out
using an ALG. I have seen some NAT routers require a re-reg interval around
15 seconds which is a bit excessing

On the issue w/ the non-working ones is you¹ll notice the rport on the one
that¹s working...

On the ones that arent working they arent specifying rport... If those are
the polycoms look for NDLB force rport setting and set it to safe... That¹s
a special mode for the polycoms...

Check the wiki for more NAT handling examples
K


On 12/5/13 8:36 PM, "Sean P Devoy" <sdevoy at bizfocused.com> wrote:

> 
> Hi all,
>  
> I seem to have fallen off the list!  No mail since 7/26/2013.  Have I made
> someone mad??  J
>  
> I have also been blissfully running with ISSUES what so ever since that time.
>  
> I have a NATed client with PERIODIC one way audio.  The client end has a Cisco
> 220W router with SIP ALG disabled.  The phones are CISCO 504Gs and Polycom
> 330s(?), basciallt the same as everywhere else we have phones.
>  
> I did a sofia global sip trace on for a while and captured the output.  After
> writing an app to sort those sip packets into logical stream files, I dug in
> to the SIP conversations.  This has left me with som questions for the people
> who know these things:
>  
> 1.   I found all my phones are REGISTERing every 23 seconds!   That seems
> absurd TO ME, so where have I screwed that up/where do I set it and what is a
> reasonable number?
> 
> 2.   The only difference I see in the SIP packets from my working phone and
> these non-working ones is of course NAT related.  The working phone¹s REGISTER
> looks like this:
> 
> recv 757 bytes from udp/[WW.XX.YY.93]:5063 at 17:53:12.164203:
> ------------------------------------------------------------------------
> REGISTER sip: <MYDOMAIN.COM> SIP/2.0
> Via: SIP/2.0/UDP WW.XX.YY.93:5063;branch=z9hG4bK-238fcc8b;rport
> 
> But the failing phones look like this:
> 
>  
> 
> recv 801 bytes from udp/[AA.BB.CC.38]:5104 at 17:53:40.693691:
> 
> ------------------------------------------------------------------------
> 
> REGISTER sip: <MYDOMAIN.COM> SIP/2.0
> 
> Via: SIP/2.0/UDP 10.2.2.239:5104;branch=z9hG4bKb374d6507894B431
> 
>  
> 
> The response is still sent to [AA.BB.CC.38]:5104
> 
>  
> I guess first, IS THAT A PROBLEM?
>  
> So, can anyone share any insight into the CISCO 220W config that I might be
> missing?  
>  
> Why would it only be a periodic problem?  Could this be a problem where one
> end is using a wider range of rports than the other supports?
>  
> Thanks,
> Sean
>  
> 
> 
> 
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-- 
Ken
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