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<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>Hi all,<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span lang=EN-GB
style='font-size:10.0pt;font-family:Arial;color:navy'>Well, one of the
interesting aspects of SRST and CME is, that they use UDP 2000 as the source
Port for all RTP traffic to locally registered phones. I believe this is
connected to the fact, that cme/srst proxies all RTP traffic from the phones to
other destination.<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span lang=EN-GB
style='font-size:10.0pt;font-family:Arial;color:navy'>e.g. if you have another
Dial-peer to a h323 system, the rtp will actually flow from the phone to udp
2000 on the router, which then reoriginates the traffic with a udp high port to
the other entity. <o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span lang=EN-GB
style='font-size:10.0pt;font-family:Arial;color:navy'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span lang=EN-GB
style='font-size:10.0pt;font-family:Arial;color:navy'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span lang=EN-GB
style='font-size:10.0pt;font-family:Arial;color:navy'>Regards<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span lang=EN-GB
style='font-size:10.0pt;font-family:Arial;color:navy'>bernhard<o:p></o:p></span></font></p>
<div>
<p style='margin-bottom:12.0pt'><font size=2 color=navy face="Times New Roman"><span
style='font-size:10.0pt;color:navy'>------------------------------------------<br>
DI (FH) Bernhard Albler, NTS AG<br>
Tel.: +43 (1) 867 67 76 -0, email: <st1:PersonName w:st="on">balbler@nts.at</st1:PersonName></span></font><o:p></o:p></p>
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</span></font></div>
<p class=MsoNormal><b><font size=2 face=Tahoma><span style='font-size:10.0pt;
font-family:Tahoma;font-weight:bold'>Von:</span></font></b><font size=2
face=Tahoma><span style='font-size:10.0pt;font-family:Tahoma'> cisco-voip-bounces@puck.nether.net
[mailto:cisco-voip-bounces@puck.nether.net] <b><span style='font-weight:bold'>Im
Auftrag von </span></b>Ed Leatherman<br>
<b><span style='font-weight:bold'>Gesendet:</span></b> Freitag, 6. Juli 2007
14:37<br>
<b><span style='font-weight:bold'>An:</span></b> Lelio Fulgenzi<br>
<b><span style='font-weight:bold'>Cc:</span></b> cisco-voip@puck.nether.net;
Matt Slaga (US)<br>
<b><span style='font-weight:bold'>Betreff:</span></b> Re: [cisco-voip] UDP 2000
from SRST router to IP phone?</span></font><o:p></o:p></p>
</div>
<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'><o:p> </o:p></span></font></p>
<p class=MsoNormal style='margin-bottom:12.0pt'><font size=3
face="Times New Roman"><span style='font-size:12.0pt'><br>
I would think that your IP phone would send to the UDP range 16384-32767 on the
gateway for RTP traffic rather than port 2000. It sounds like the call is
already setup.<br>
<br>
<o:p></o:p></span></font></p>
<div>
<p class=MsoNormal><span class=gmailquote><font size=3 face="Times New Roman"><span
style='font-size:12.0pt'>On 7/5/07, <b><span style='font-weight:bold'>Lelio
Fulgenzi</span></b> <<a href="mailto:lelio@uoguelph.ca">lelio@uoguelph.ca</a>>
wrote:</span></font></span><o:p></o:p></p>
<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'>That's was our understanding, TCP 2000.<br>
<br>
I don't think we've ever heard about the sequential ports though. We've only<br>
opened up TCP 2000.<br>
<br>
----- Original Message -----<br>
From: "Matt Slaga (US)" < <a href="mailto:Matt.Slaga@us.didata.com">Matt.Slaga@us.didata.com</a>><br>
To: "Paul Choi" <<a href="mailto:asobihoudai@yahoo.com">asobihoudai@yahoo.com</a>>;
"Lelio Fulgenzi"<br>
<<a href="mailto:lelio@uoguelph.ca"> lelio@uoguelph.ca</a>>; <<a
href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a>><br>
Sent: Thursday, July 05, 2007 9:09 PM<br>
Subject: RE: [cisco-voip] UDP 2000 from SRST router to IP phone?<br>
<br>
<br>
> Cisco contends that SCCP uses TCP 2000 - 2002 (one port for each<br>
> CallManager in it's failover list). Not sure about UDP 2000.<br>
><br>
> -----Original Message-----<br>
> From: <a href="mailto:cisco-voip-bounces@puck.nether.net">cisco-voip-bounces@puck.nether.net</a><br>
> [mailto:<a href="mailto:cisco-voip-bounces@puck.nether.net">cisco-voip-bounces@puck.nether.net</a>]
On Behalf Of Paul Choi<br>
> Sent: Thursday, July 05, 2007 9:01 PM<br>
> To: Lelio Fulgenzi; <a href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a><br>
> Subject: Re: [cisco-voip] UDP 2000 from SRST router to IP phone?<br>
><br>
> I could be full of shit but I'm pretty sure that SCCP <br>
> uses UDP 2000 to communicate with CCM.<br>
><br>
><br>
> --- Lelio Fulgenzi <<a href="mailto:lelio@uoguelph.ca">lelio@uoguelph.ca</a>>
wrote:<br>
><br>
>> We're just setting up some ACLs and found that while <br>
>> in SRST mode, the SRST router (or at least the<br>
>> loopback address that we bind MGCP to) tried to<br>
>> communicate to the IP phone that was registered. We<br>
>> had not allowed for this and as a result, the call <br>
>> out the gateway was one way audio. But we didn't see<br>
>> any other denies come out of the logs.<br>
>><br>
>> I see that in the SRST config, it asks for a port,<br>
>> and it is 2000. But it's still wierd to see UDP 2000 <br>
>> being used.<br>
>><br>
>> Can anyone comment?<br>
>><br>
>> 2d08h: %SEC-6-IPACCESSLOGP: list voice_in_kc denied<br>
>> udp 10.104.122.141(17624) -> 10.104.34.48(2000), 1<br>
>> packet <br>
>><br>
>> ^- IP
phone ^-
SRST router<br>
>><br>
>><br>
>><br>
> ------------------------------------------------------------------------<br>
> --------<br>
>> Lelio Fulgenzi, B.A.<br>
>> Senior Analyst (CCS) * University of Guelph *<br>
>> Guelph, Ontario N1G 2W1<br>
>> (519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)<br>
>><br>
> ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ <br>
>><br>
>> "Buffalo buffalo Buffalo buffalo buffalo buffalo<br>
>> Buffalo buffalo." WJR<br>
>> > _______________________________________________<br>
>> cisco-voip mailing list <br>
>> <a href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a><br>
>> <a href="https://puck.nether.net/mailman/listinfo/cisco-voip">https://puck.nether.net/mailman/listinfo/cisco-voip</a><br>
>><br>
><br>
><br>
><br>
><br>
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<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'><br>
<br clear=all>
<br>
-- <br>
Ed Leatherman<br>
Senior Voice Engineer<br>
West Virginia University<br>
Telecommunications and Network Operations <o:p></o:p></span></font></p>
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