SIP Trace Question / NAT one way audio problem.
Hi all, I seem to have fallen off the list! No mail since 7/26/2013. Have I made someone mad?? :) 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 --- This email is free from viruses and malware because avast! Antivirus protection is active. http://www.avast.com
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Sean P Devoy