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<TITLE>Re: [Freeswitch-users] SIP Trace Question / NAT one way audio problem.</TITLE>
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<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>
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On 12/5/13 8:36 PM, "Sean P Devoy" <<a href="sdevoy@bizfocused.com">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: <MYDOMAIN.COM> 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: <MYDOMAIN.COM> 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>
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