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

[Michael Jerris]

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…