Unless you put a border element in, plan to see traffic between phone-to-phone, cucm-to-phone, and phone-to-cucm on to/from both sides. If you put a CUBE in, each side talks to a single IP address on their side, thus 2 IP addresses. If you have overlapping IP addresses CUBE is by far the preferred choice.<br>
<br>-nick<br><br><br><br><div class="gmail_quote">On Mon, Sep 27, 2010 at 1:24 AM, Humayun Sami <span dir="ltr"><<a href="mailto:humayun_sami@hotmail.com">humayun_sami@hotmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt 0.8ex; border-left: 1px solid rgb(204, 204, 204); padding-left: 1ex;">
<div>
<p class="MsoNormal"><span style="font-size: 10pt; font-family: "Trebuchet MS"; color: navy;">Thank everyone, I have a question here, lets say that we can get it
to work with using NAT. I wanted to understand the design that how does the IP
to IP (Call Manager to Manager) and phone to phone connectivity works. As in my
knowledge we have separate IP’s for Call Manager. My question is in respect to
the RTP session establishment.</span></p>
<p class="MsoNormal"><span style="font-size: 10pt; font-family: "Trebuchet MS"; color: navy;"> </span></p>
<p class="MsoNormal"><span style="font-size: 10pt; font-family: "Trebuchet MS"; color: navy;">I understand it with that we can provide NAT’d addresses to Call
Managers, how will the Phone work here. Even if we have NAT’d the phones subnet
as well. To create RTP session phones will definitely have direct session
established. Can you please make me understand the design working here in
regards to the phone communication and RTP session establishment.<br><br> Regards,<br> Humayun Sami<br></span></p><p class="MsoNormal"><br></p><p class="MsoNormal"><br></p><p class="MsoNormal"><br></p><p class="MsoNormal">
<br><span style="font-size: 10pt; font-family: "Trebuchet MS"; color: navy;"></span></p>
<hr>Date: Sun, 26 Sep 2010 11:38:43 -0700<br>From: <a href="mailto:nice_chatin@yahoo.com" target="_blank">nice_chatin@yahoo.com</a><br>To: <a href="mailto:ahmed_elnagar@rayacorp.com" target="_blank">ahmed_elnagar@rayacorp.com</a>; <a href="mailto:matthnick@gmail.com" target="_blank">matthnick@gmail.com</a><br>
CC: <a href="mailto:cisco-voip@puck.nether.net" target="_blank">cisco-voip@puck.nether.net</a><div><div></div><div class="h5"><br>Subject: Re: [cisco-voip] NAT for Call Managers<br><br><table border="0" cellpadding="0" cellspacing="0">
<tbody><tr><td style="font: inherit;" valign="top">Put a Session Border Controller (IPIP Gateway) in between.<br><br>--- On <b>Sat, 9/25/10, Nick Matthews <i><<a href="mailto:matthnick@gmail.com" target="_blank">matthnick@gmail.com</a>></i></b> wrote:<br>
<blockquote style="margin-left: 5px; padding-left: 5px;"><br>From: Nick Matthews <<a href="mailto:matthnick@gmail.com" target="_blank">matthnick@gmail.com</a>><br>Subject: Re: [cisco-voip] NAT for Call Managers<br>To: "Ahmed Elnagar" <<a href="mailto:ahmed_elnagar@rayacorp.com" target="_blank">ahmed_elnagar@rayacorp.com</a>><br>
Cc: <a href="mailto:cisco-voip@puck.nether.net" target="_blank">cisco-voip@puck.nether.net</a><br>Date: Saturday, September 25, 2010, 4:33 AM<br><br><div>You can get it to work, but historically it's one of the more difficult things to do. Both SIP and H.323 have their various problems with NAT. With H323, it uses a random port<->random port exchange which firewalls don't like. Some H.323 endpoints may use non-standard extensions/fields. With SIP, it routinely includes IP
addresses in fields that should and should not be NAT'd, and often firewalls don't respect the right ones (or at all). <br>
<br>For instance, the newer phone loads use SCCP v17 which caused some firewalls to stop recognizing the SCCP format. So, tread carefully is the answer. <br><br>I think RTP is generally handled pretty well, but it's usually the signaling protocols that get ignored. Make sure you know what they are and try to configure the firewall to be aware of them.<br>
<br>-nick<br><br><div>On Fri, Sep 24, 2010 at 4:24 PM, Ahmed Elnagar <span dir="ltr"><<a rel="nofollow" href="http:///mc/compose?to=ahmed_elnagar@rayacorp.com" target="_blank">ahmed_elnagar@rayacorp.com</a>></span> wrote:<br>
<blockquote style="padding-left: 1ex;">
<div lang="EN-US">
<div>
<p><span style="font-size: 11pt; color: rgb(31, 73, 125);">NAT have a lot of problems one of them that I faced myself that
the phones show as register on CUCM but actually they are not…try to avoid it
as much as you can…however if it is mandatory choose the device that is making
NAT for VOIP traffic VOIP aware in order to rewrite the NATted packet in the
correct way that is suitable for RTP.</span></p>
<p><span style="font-size: 11pt; color: rgb(31, 73, 125);"> </span></p>
<div>
<div>
<p style="background: none repeat scroll 0% 0% white;"><span style="font-size: 10pt; color: rgb(31, 73, 125);"> </span><span style="font-size: 10.5pt; color: rgb(31, 73, 125);">Best
Regards;</span></p>
<p style="background: none repeat scroll 0% 0% white;"><span style="font-size: 10.5pt; color: rgb(31, 73, 125);"> Ahmed Elnagar</span></p>
<p style="background: none repeat scroll 0% 0% white;"><span style="font-size: 10.5pt; color: rgb(31, 73, 125);"> Senior Network PS
Engineer</span></p>
<p style="background: none repeat scroll 0% 0% white;"><span style="font-size: 10.5pt; color: rgb(31, 73, 125);"> Mob: +2019-0016211</span></p>
<p style="background: none repeat scroll 0% 0% white;"><span style="font-size: 10.5pt; color: rgb(31, 73, 125);"> CCIE#24697 (Voice)</span></p>
<p style="background: none repeat scroll 0% 0% white;"><span style="font-size: 10.5pt; color: rgb(31, 73, 125);"> <img src="cid:1.1338612220@web55004.mail.re4.yahoo.com" alt="ccie_voice_large.gif" height="63" width="63"></span></p>
</div>
</div>
<p><span style="font-size: 11pt; color: rgb(31, 73, 125);"> </span></p>
<div>
<div style="border-width: 1pt medium medium; border-style: solid none none; border-color: rgb(181, 196, 223); padding: 3pt 0in 0in;">
<p><b><span style="font-size: 10pt;">From:</span></b><span style="font-size: 10pt;">
<a rel="nofollow" href="http:///mc/compose?to=cisco-voip-bounces@puck.nether.net" target="_blank">cisco-voip-bounces@puck.nether.net</a> [mailto:<a rel="nofollow" href="http:///mc/compose?to=cisco-voip-bounces@puck.nether.net" target="_blank">cisco-voip-bounces@puck.nether.net</a>] <b>On
Behalf Of </b><a rel="nofollow" href="http:///mc/compose?to=john_burk@oxy.com" target="_blank">john_burk@oxy.com</a><br>
<b>Sent:</b> Friday, September 24, 2010 4:57 PM<br>
<b>To:</b> <a rel="nofollow" href="http:///mc/compose?to=humayun_sami@hotmail.com" target="_blank">humayun_sami@hotmail.com</a>; <a rel="nofollow" href="http:///mc/compose?to=cisco-voip@puck.nether.net" target="_blank">cisco-voip@puck.nether.net</a><br>
<b>Subject:</b> Re: [cisco-voip] NAT for Call Managers</span></p>
</div>
</div><div><div></div><div>
<p> </p>
<div>
<p><span style="font-size: 10pt; color: navy;">NAT can cause havoc with the RTP stream, but with the right
firewall and design/config it can be made to work. I use VPN if at all possible
to avoid these issues.<br>
<br>
John Burk, Consultant <br>
Sent from my Blackberry</span><span style="font-size: 10pt;"></span></p>
</div>
<p><span style="font-size: 10pt;"> </span></p>
<div>
<div style="text-align: center;" align="center"><span style="font-size: 10pt;">
<hr align="center" size="2" width="100%">
</span></div>
<p style="margin-bottom: 12pt;"><b><span style="font-size: 10pt;">From</span></b><span style="font-size: 10pt;">: <a rel="nofollow" href="http:///mc/compose?to=cisco-voip-bounces@puck.nether.net" target="_blank">cisco-voip-bounces@puck.nether.net</a>
<<a rel="nofollow" href="http:///mc/compose?to=cisco-voip-bounces@puck.nether.net" target="_blank">cisco-voip-bounces@puck.nether.net</a>> <br>
<b>To</b>: <a rel="nofollow" href="http:///mc/compose?to=cisco-voip@puck.nether.net" target="_blank">cisco-voip@puck.nether.net</a> <<a rel="nofollow" href="http:///mc/compose?to=cisco-voip@puck.nether.net" target="_blank">cisco-voip@puck.nether.net</a>> <br>
<b>Sent</b>: Fri Sep 24 08:02:38 2010<br>
<b>Subject</b>: [cisco-voip] NAT for Call Managers </span></p>
</div>
<p style="margin-bottom: 12pt;"><span style="font-size: 10pt;">Is it recommended to use NAT on VoIP. I have
two separate cluster one for cisco call manager and other for Avaya. We are
integrating both the setups(h323). Is it okay to use NAT. Can someone provide
me a document which helps in this reference.<br>
<br>
<br>
For what reason people do not recommend it in the network, or is it the same
that you do not register your call managers with the domain server. Look
forward to hear.<br>
<br>
<br>
<br>
Regards,<br>
Humayun Sami.</span></p>
</div></div></div>
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