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

Avi Marcus avi at avimarcus.net
Fri Dec 6 06:54:33 MSK 2013


I set the linksys spas to have a ping of 15 seconds, but not to re register
every 15 seconds!

When I see constant register attempts, it always means that the 401 was not
received so the register never completes. Are these phones actually
successfully registered?

-Avi
On Dec 6, 2013 4:47 AM, "Ken Rice" <krice at freeswitch.org> wrote:

>  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 you’ll notice the rport on the one
> that’s 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... That’s
> 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 phone’s
> 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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