[Freeswitch-users] Simple IAX2 setup - help with converting from asterisk to freeswitch

[Anthony Minessale]

Well what IAX gives us on FreeSWITCH is random segfaults because the
protocol changed in a way
that made the IAX stack library we were using a deathtrap.  The authors of
IAX decided to change the protocol
so that it was not backwards compatible with their own previous work and due
to lack of resources we decided that removing it was the best way
to achieve stability.

IAX offers some nat relief with the 1 port strategy at the expense of the
stack implementation in user space.
Frankly the authors of this protocol failed to supply a robust reference
implementation of the protocol in a standalone license free package.  There
was one small library designed specifically for a standalone client (that's
the one we used) and as I explained, in the end, it failed us.

It then boils down to the business question: Do I want to chase after
marginal savings on my configuration and packet sizes at the expense of much
fewer calls and a serious risk of catastrophic failure caused by a poor
implementation?

SIP has its flaws as well, we don't have enough time to get into that, but
on FreeSWITCH it can scale much higher than IAX even when it was working
better (before they changed it).

You said you are new to VoIP (which is surprising based on your email addr)
so from your perspective what is it that you are looking for?

rtp packetization refers to sending more audio at once to reduce the
overhead of IP headers.
in FreeSWITCH you can configure your codecs to PCMU at 60i for instance to
bundle 3 typical packets into 1

BTW, No matter what you use, you will have little luck on a virtual machine
as your first experience.
I recommend you get a real box (multi-core 64bit) to play with.


On Wed, May 26, 2010 at 3:19 AM, Michael Toop <
michaelt at voxcore.voxtelecom.co.za> wrote:

> Hi,
>
>  Can someone assist me with how one would achieve what Michael Jerris
> suggested: "IAX offers less overhead on rtp, but the same thing can be
> accomplished with rtp using packetization..."   ...?
>
>  I have done tons of reading on cRTP etc & I must be missing something as
> there is no obvious stock standard Linux way to achieve what IAX give you?
>
> Kind Regards,
>
> Michael
>
>
> On Fri, Feb 5, 2010 at 5:54 PM, Michael Jerris <mike at jerris.com> wrote:
>
>> I find the secure and efficiency claims on IAX to be pretty much a farce.
>>  IAX offers less overhead on rtp, but the same thing can be accomplished
>> with rtp using packetization, as for security, I don't see any credible
>> claim on that.  IAX also forces the program to sort out a ton of audio for
>> different users going to 1 socket, something that a network stack is quite
>> good at when using different ports, but is a lot more work where we are
>> getting the packets.  As for default passwords and users, of course I
>> wouldn't use those in production, those are for you to see how the pieces
>> work together out of the box.  I wouldn't however quickly scrap the entire
>> default config, just read through them and think about what you need and do
>> not.  The extension ranges you use is totally at your discretion.
>>
>> Mike
>>
>>
>> On Feb 5, 2010, at 8:53 AM, Matthew Law wrote:
>>
>> > Why is that? - a lot of web pages I have read claim that IAX is more
>> > secure and efficient.  I have no problem with using SIP whatsoever and
>> it
>> > certainly appears to be ubiquitous.  I am a complete newcomer to VoIP
>> and
>> > I am trying to do this as securely as possible since I want to run
>> > freeswitch on a Xen VPS which will be visible on the internet.
>> >
>> > Anyway, since my first question, I have worked my way through the wiki,
>> > read a lot of example configs and made some sense of the docs.  I now
>> have
>> > a very basic config working (with SIP) on a local vmware image using the
>> > 'quick and dirty' Makefile method.  I removed all of the example configs
>> > from the xml files (those default extensions and passwords scared me)
>> and
>> > added just one for extension 1000, plus my dialplan and inbound/outbound
>> > settings.
>> >
>> > One question: is there any reason not to use a smaller extension number
>> > range, like 200-210, for example?
>> >
>> > Now to figure out how to get time based roaming working…
>>
>>
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>
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-- 
Anthony Minessale II

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