Hi,<div><br></div><div> 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..." ...?</div>
<div><br></div><div> 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?</div><div><br></div><div>Kind Regards,</div><div>
<br>Michael</div><div> <br><br><div class="gmail_quote">On Fri, Feb 5, 2010 at 5:54 PM, Michael Jerris <span dir="ltr"><<a href="mailto:mike@jerris.com">mike@jerris.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">
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.<br>
<br>
Mike<br>
<div class="im"><br>
<br>
On Feb 5, 2010, at 8:53 AM, Matthew Law wrote:<br>
<br>
> Why is that? - a lot of web pages I have read claim that IAX is more<br>
> secure and efficient. I have no problem with using SIP whatsoever and it<br>
> certainly appears to be ubiquitous. I am a complete newcomer to VoIP and<br>
> I am trying to do this as securely as possible since I want to run<br>
> freeswitch on a Xen VPS which will be visible on the internet.<br>
><br>
> Anyway, since my first question, I have worked my way through the wiki,<br>
> read a lot of example configs and made some sense of the docs. I now have<br>
> a very basic config working (with SIP) on a local vmware image using the<br>
> 'quick and dirty' Makefile method. I removed all of the example configs<br>
> from the xml files (those default extensions and passwords scared me) and<br>
> added just one for extension 1000, plus my dialplan and inbound/outbound<br>
> settings.<br>
><br>
> One question: is there any reason not to use a smaller extension number<br>
> range, like 200-210, for example?<br>
><br>
</div>> Now to figure out how to get time based roaming working…<br>
<div><div></div><div class="h5"><br>
<br>
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