<p>I set the linksys spas to have a ping of 15 seconds, but not to re register every 15 seconds!</p>
<p>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?</p>
<p>-Avi</p>
<div class="gmail_quote">On Dec 6, 2013 4:47 AM, "Ken Rice" <<a href="mailto:krice@freeswitch.org">krice@freeswitch.org</a>> wrote:<br type="attribution"><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div>
<font face="Monaco, Courier New"><span style="font-size:11pt">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<br>
<br>
On the issue w/ the non-working ones is you’ll notice the rport on the one that’s working...<br>
<br>
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...<br>
<br>
Check the wiki for more NAT handling examples<br>
K<br>
<br>
<br>
On 12/5/13 8:36 PM, "Sean P Devoy" <<a href="http://sdevoy@bizfocused.com" target="_blank">sdevoy@bizfocused.com</a>> wrote:<br>
<br>
</span></font><blockquote><font face="Monaco, Courier New"><span style="font-size:11pt"><img src="cid:3469121262_7318247"><br>
</span></font><font color="#44546A"><font face="Verdana, Helvetica, Arial"><span style="font-size:12pt">Hi all,<br>
<br>
I seem to have fallen off the list! No mail since 7/26/2013. Have I made someone mad?? </span></font><span style="font-size:12pt"><font face="Wingdings">J<br>
</font><font face="Verdana, Helvetica, Arial"> <br>
I have also been blissfully running with ISSUES what so ever since that time.<br>
<br>
I have a NATed client with <b>PERIODIC</b> 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.<br>
<br>
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:<br>
<br>
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?<br>
</font></span></font><font face="Monaco, Courier New"><span style="font-size:11pt"><br>
</span></font><font color="#44546A"><font face="Verdana, Helvetica, Arial"><span style="font-size:12pt">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:<br>
<br>
recv 757 bytes from udp/[WW.XX.YY.93]:5063 at 17:53:12.164203:<br>
------------------------------------------------------------------------ <br>
REGISTER sip: <<a href="http://MYDOMAIN.COM" target="_blank">MYDOMAIN.COM</a>> SIP/2.0<br>
Via: SIP/2.0/UDP WW.XX.YY.93:5063;branch=z9hG4bK-238fcc8b;rport<br>
<br>
But the failing phones look like this:<br>
</span></font></font><font face="Monaco, Courier New"><span style="font-size:11pt"><br>
</span></font><font color="#44546A"><font face="Verdana, Helvetica, Arial"><span style="font-size:12pt"> <br>
</span></font></font><font face="Monaco, Courier New"><span style="font-size:11pt"><br>
</span></font><font color="#44546A"><font face="Verdana, Helvetica, Arial"><span style="font-size:12pt">recv 801 bytes from udp/[AA.BB.CC.38]:5104 at 17:53:40.693691:<br>
</span></font></font><font face="Monaco, Courier New"><span style="font-size:11pt"><br>
</span></font><font color="#44546A"><font face="Verdana, Helvetica, Arial"><span style="font-size:12pt">------------------------------------------------------------------------<br>
</span></font></font><font face="Monaco, Courier New"><span style="font-size:11pt"><br>
</span></font><font color="#44546A"><font face="Verdana, Helvetica, Arial"><span style="font-size:12pt">REGISTER sip: <<a href="http://MYDOMAIN.COM" target="_blank">MYDOMAIN.COM</a>> SIP/2.0<br>
</span></font></font><font face="Monaco, Courier New"><span style="font-size:11pt"><br>
</span></font><font color="#44546A"><font face="Verdana, Helvetica, Arial"><span style="font-size:12pt">Via: SIP/2.0/UDP 10.2.2.239:5104;branch=z9hG4bKb374d6507894B431<br>
</span></font></font><font face="Monaco, Courier New"><span style="font-size:11pt"><br>
</span></font><font color="#44546A"><font face="Verdana, Helvetica, Arial"><span style="font-size:12pt"> <br>
</span></font></font><font face="Monaco, Courier New"><span style="font-size:11pt"><br>
</span></font><font color="#44546A"><font face="Verdana, Helvetica, Arial"><span style="font-size:12pt">The response is still sent to [AA.BB.CC.38]:5104<br>
</span></font></font><font face="Monaco, Courier New"><span style="font-size:11pt"><br>
</span></font><font color="#44546A"><font face="Verdana, Helvetica, Arial"><span style="font-size:12pt"> <br>
I guess first, IS THAT A PROBLEM?<br>
<br>
So, can anyone share any insight into the CISCO 220W config that I might be missing? <br>
<br>
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?<br>
<br>
Thanks,<br>
Sean<br>
<br>
</span></font></font><font face="Monaco, Courier New"><span style="font-size:11pt"><br>
<br>
<hr align="CENTER" size="3" width="100%"> <<a href="http://www.avast.com/" target="_blank">http://www.avast.com/</a>> </span></font><font color="#3D4D5A"><font face="Calibri, Verdana, Helvetica, Arial"><span style="font-size:12pt">This email is free from viruses and malware because avast! Antivirus <<a href="http://www.avast.com/" target="_blank">http://www.avast.com/</a>> protection is active. <br>
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