[asterisk-users] [IAX] Recommended soft- and hardphones?

[Michelle Dupuis]

Because RTP ports are assigned dynamically (and not necessarily
symmetrically) during call setup using SIP, you need a SIP aware firewall.
Without one, you may get SIP registration, but usually one-way/no audio
(RTP).

Most hotels and hotspots do NOT support SIP - either because they run cheap
firewalls/routers or because VoIP competes with other services they offer.
IAX is single port and symmetrical so even cheap firewalls/routers can pass
this without additional setup.

Our consultants travel across North America and we finally gave up on SIP
phones because of these hassles.  (Trying to explain SIP, NAT, IP
Masquerading, symmetry, RTP, etc to tech support for each hotel was a time
waster.  *We* know what the NAT/IP Masquerading issue is - but that doesn't
help some tech support guy in India assisting Marriott customers).  Perhaps
wherever you are located the state of firewalls/routers is different.


-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Alex Balashov
Sent: Friday, October 30, 2009 8:37 AM
To: Asterisk Users List
Subject: Re: [asterisk-users] [IAX] Recommended soft- and hardphones?

My experience does not support your conclusions.  In my personal
observations of situations in which I have been involved, most allegations
of serious SIP problems related to source NAT ("IP
masquerading") are exaggerations stemming from lack of subject matter
comprehension.  This is bearing in mind, duly, that SIP and NAT *is*,
inherently, a problematic equation - of that, there can be no question.

But I've never had problems getting a SIP soft phone to make and receive
calls from anywhere I've taken it.  The only substantial problem I've run
into is that many NAT gateways lose UDP state quite quickly, so after a
certain period of inactivity, calls cannot be received;  this is solved by
decreasing the re-registration interval, or increasing the frequency of
state-sustaining SIP OPTIONS pings, etc.  Many service providers have
implemented such steps since the last time I was involved with this problem
seriously.

I'll take your word for the fact that IAX may be easier, though.

Michelle Dupuis wrote:

> I assume you're kidding?!
> 
> RTP is mangled/blocked by most hotspots and mid-size company firewalls...
> 
> IAX is often the only way our staff can connect while on the road.
> 
> -----Original Message-----
> From: asterisk-users-bounces at lists.digium.com
> [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Alex 
> Balashov
> Sent: Friday, October 30, 2009 8:03 AM
> To: Asterisk Users List
> Subject: Re: [asterisk-users] [IAX] Recommended soft- and hardphones?
> 
> Vincent wrote:
> 
>> Since SIP/RTP is a pain to use with road warriors who need to connect 
>> from any location over the Internet, I'd like to get them some IAX 
>> phones instead.
> 
> What gives you that idea?
> 
> --
> Alex Balashov - Principal
> Evariste Systems
> Web     : http://www.evaristesys.com/
> Tel     : (+1) (678) 954-0670
> Direct  : (+1) (678) 954-0671
> 
> _______________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> 
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
> 
> 
> _______________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> 
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users


--
Alex Balashov - Principal
Evariste Systems
Web     : http://www.evaristesys.com/
Tel     : (+1) (678) 954-0670
Direct  : (+1) (678) 954-0671

_______________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users