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

Alex Balashov abalashov at evaristesys.com
Fri Oct 30 08:14:55 CDT 2009


On the contrary, SIP-aware ALGs mostly do more harm than good. While  
their purpose is noble, their implementation is frequently lacking/ 
incomplete and conflicts with existing far-end NAT traversal  
approaches taken by most service providers these days.  These are also  
present in commercial PBX equipment, e.g. along the lines of nat=yes  
in sip.conf.

Almost all UAs symmetrically signal and the overwhelming majority do  
symmetrical RTP as well.  This used to be more of a problem a few  
years ago.

I don't disagree that IAX may be easier still.  I just haven't found  
that the alleged problems of SIP and NAT live up to the hype now that  
almost every commercial service edge does far-end NAT detection and  
draft-comedia style RTP handling.

--
Sent from mobile device

On Oct 30, 2009, at 8:58 AM, Michelle Dupuis <support at ocg.ca> wrote:

> 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
>>
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>
> --
> 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:
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>
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