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

[Michael Toop]

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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