[cisco-voip] NAT for Call Managers

Mon Sep 27 09:40:17 EDT 2010

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.

-nick

On Mon, Sep 27, 2010 at 1:24 AM, Humayun Sami <humayun_sami at hotmail.com>wrote:

>  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.
>
>
>
> 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.
>
> Regards,
> Humayun Sami
>
>
>
>
>
> ------------------------------
> Date: Sun, 26 Sep 2010 11:38:43 -0700
> From: nice_chatin at yahoo.com
> To: ahmed_elnagar at rayacorp.com; matthnick at gmail.com
> CC: cisco-voip at puck.nether.net
>
> Subject: Re: [cisco-voip] NAT for Call Managers
>
> Put a Session Border Controller (IPIP Gateway) in between.
>
> --- On *Sat, 9/25/10, Nick Matthews <matthnick at gmail.com>* wrote:
>
>
> From: Nick Matthews <matthnick at gmail.com>
> Subject: Re: [cisco-voip] NAT for Call Managers
> To: "Ahmed Elnagar" <ahmed_elnagar at rayacorp.com>
> Cc: cisco-voip at puck.nether.net
> Date: Saturday, September 25, 2010, 4:33 AM
>
> 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).
>
> 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.
>
> 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.
>
> -nick
>
> On Fri, Sep 24, 2010 at 4:24 PM, Ahmed Elnagar <ahmed_elnagar at rayacorp.com<http:///mc/compose?to=ahmed_elnagar@rayacorp.com>
> > wrote:
>
>  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.
>
>
>
>  Best Regards;
>
>   Ahmed Elnagar
>
>   Senior Network PS Engineer
>
>   Mob: +2019-0016211
>
>   CCIE#24697 (Voice)
>
>  [image: ccie_voice_large.gif]
>
>
>
> *From:* cisco-voip-bounces at puck.nether.net<http:///mc/compose?to=cisco-voip-bounces@puck.nether.net>[mailto:
> cisco-voip-bounces at puck.nether.net<http:///mc/compose?to=cisco-voip-bounces@puck.nether.net>]
> *On Behalf Of *john_burk at oxy.com <http:///mc/compose?to=john_burk@oxy.com>
> *Sent:* Friday, September 24, 2010 4:57 PM
> *To:* humayun_sami at hotmail.com<http:///mc/compose?to=humayun_sami@hotmail.com>;
> cisco-voip at puck.nether.net<http:///mc/compose?to=cisco-voip@puck.nether.net>
> *Subject:* Re: [cisco-voip] NAT for Call Managers
>
>
>
> 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.
>
> John Burk, Consultant
> Sent from my Blackberry
>
>
>  ------------------------------
>
> *From*: cisco-voip-bounces at puck.nether.net<http:///mc/compose?to=cisco-voip-bounces@puck.nether.net><
> cisco-voip-bounces at puck.nether.net<http:///mc/compose?to=cisco-voip-bounces@puck.nether.net>>
>
> *To*: cisco-voip at puck.nether.net<http:///mc/compose?to=cisco-voip@puck.nether.net><
> cisco-voip at puck.nether.net<http:///mc/compose?to=cisco-voip@puck.nether.net>>
>
> *Sent*: Fri Sep 24 08:02:38 2010
> *Subject*: [cisco-voip] NAT for Call Managers
>
> 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.
>
>
> 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.
>
>
>
> Regards,
> Humayun Sami.
>
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