[asterisk-users] Same provider - IAX sounds bad, SIP sounds great

[Steve Totaro]

On Wed, Feb 29, 2012 at 9:22 AM, Alejandro Imass <ait at p2ee.org> wrote:

> On Tue, Feb 28, 2012 at 8:58 PM, Steve Totaro
> <stotaro at asteriskhelpdesk.com> wrote:
> >
> >
>
> [...]
> > Yes, I have had no problems with Grandstream first gen ATAs, configured
> with
> > server and credentials and shipped off, they just work.
>
> We use the HT-286, the server is on a public IP the nat setting on
> asterisk is set to yes and without port re-direction the ATAs have
> never connected from a private network, so I honestly find this "SIP
> plug and play" very hard to believe. But if it is true, then maybe you
> can actually help us figure out all the NAT issues we've had with SIP
> for the past 5 years. Perhaps, it is simply ignorance on our side and
> we have something fundamentally wrong in our set-up somewhere that may
> be have been causing these issues with NAT.
>
> Our set-up is fundamentally public and private Asterisk servers
> running on FreeBSD. Versions may vary from FBSD 7 thru 8.2 and
> Asterisk 1.4 and 1.6. We are planning to upgrade every server to FBSD
> 8.2 and Asterisk 1.8 but we are in that process right now. Some
> Asterisk run in jails so I can understand the NAT issues there may be
> caused by the server itself. I honestly *love* your OpenVPN idea but I
> have to find a cheap ATA that could run as an OpenVPN client.
>
> Taking the simplest example a simple Asterisk 1.6 server on a public
> IP running on the base system (not in a jail):
>
> We run an operation that spans several countries including Canada, the
> US and the Latin American Andean region. As examples, with Canadian
> ISPs such as Rogers and Bell  we have always had to redirect the ports
> and use STUN server for the HT-286 to register to the Asterisk server.
>
> In the US we have the same problem with Comcast networks, so I don't
> understand how you say that you plug a Grandtream SIP ATA to a Comcast
> router and it just works. However, in a couple of NOLA countries the
> ISP's routers actually give public IPs, so if the SIP ATAs are
> connected directly to the ISP router, or in the DMZ then it just works
> as you say, BUT if the ATA is connected behind the firewall, or to a
> WiFi router, then we've _allways_  had to redirect ports. In every
> sigle customer we have had to send instructions on how to redirect
> ports, and of course to configure firewall if present.
>
> I just don't understand how you and other here say that a SIP ATA can
> "just work". On the contrarty, with IAX2 using cheap AG-188N from
> Atcom they are just plug and play when shipped with a standard conf,
> and we have none of the quality issues you are referring to. We do
> have some call drops however, and some hangup problems but they don't
> affect our clients as much as having to deal with NAT issues.
>
> We may not run 15K extensions like you but I think we have a pretty
> good testing ground and have dealt with a fair share of NAT problems
> with SIP, that you and others here apparently don't have, so I am as
> amazed by your likeness of SIP as perhaps you are amazed as our
> likeness of IAX.
>
>
If you can post some SIP debug info from an ATA trying to register without
any redirection and also the relevant portions of your sip.conf, I am sure
I can help.

Do it from a new location with an el cheapo home router, Linksys WRTXXX.

If I cannot help you in a few emails, we can take this offline.

Actually paste your entire sip.conf in pastebin or something, as well as
sip debug.

Also the configs of your ATAs.

I think you have over-engineered to the point of creating problems.  This
is very common.  My philosophy is "KISS"

Thanks,
Steve T
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