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@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
<http://www.avast.com/> This email is free from viruses and malware because avast! Antivirus <http://www.avast.com/> protection is active.
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org G+ ClueCon : http://fs0.us/cluecon-gplus FB ClueCon : http://fs0.us/cluecon-fb G+ FreeSwitch : http://fs0.us/freeswitch-gplus FB FreeSWITCH : http://fs0.us/freeswitch-fb Twitter : @FreeSWITCH_WIRE irc.freenode.net #freeswitch