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

[Alejandro Imass]

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.