On Thu, May 13, 2010 at 4:17 AM, Vieri <span dir="ltr"><<a href="mailto:rentorbuy@yahoo.com">rentorbuy@yahoo.com</a>></span> wrote:<br><div class="gmail_quote"><blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt 0.8ex; border-left: 1px solid rgb(204, 204, 204); padding-left: 1ex;">
Hi,<br>
<br>
I have an audio quality problem regarding IAX2. I have 2 Asterisk servers interconnected via 2 LAN trunks at 1Gbps (no nat, no firewall).<br>
One trunk is SIP and the other IAX2.<br>
Normally, I use IAX2 but have noticed easily reproducible audio quality problems (voice in/out is OK but there's a "third" noise overlapping with a "scratchy sound" as if it were some kind of interference).<br>
<br>
So lately I setup calls to go through the SIP trunk and audio quality is OK (no "third overlapping noise").<br>
<br>
This is happening between Asterisk 1.4.31 and a 1.2.40.<br>
<br>
I'm wondering if there's something I can tweak in IAX2 to eliminate this artifact.<br>
<br>
Could the IAX2 jitter buffer between 1.2 and 1.4 be an issue (I believe it's enabled by default)?<br>
<br>
Thanks,<br>
<font color="#888888"><br>
Vieri<br>
</font><div><div></div><div class="h5"><br><br></div></div></blockquote>IAX2 has been borken since Jump Street. You can find posts of mine
dating back years stating exactly what you just described. Of course,
plenty of people have "No Problems" with IAX2, and the official Digium
party line is that is "Great" <br>
<br>
I only use it for situations where NAT makes SIP impossible since it
only uses one port for everything. Don't bother with it on a LAN or
VPN, only use it in a pinch.<br>
<br>
I suggest that you use 1.4.X or newer if you 1.6.X if you are a bit
daring. I believe IAX2 has had alot of reworking, so using the latest
and same version on your boxen should help, but although Asterisk is not
RFC SIP compliant, it works well. IAX2 looks and sound good on paper,
just not the every important phone.<br>
<br>
I have made a good deal of money consulting, only to find out that the
customer was using IAX2 somewhere, after switching to SIP, they had
perfect audio, these were mostly ITSPs using "trunking" but I have seen
the same issue with no trunking (using same protocol overhead for all
simultaneous calls, rather than a separate overhead per call.<br>
<br>
I advise bagging IAX2 for now if you can. I am forced to use it when
ISPs are using several NATs in their networks, but then and only then.
If you can setup OpenVPN, then do it and use SIP. If you are forced to
use IAX2, don't trunk and try to use the latest stable, or a version or
two prior.<br><br>I have a few friends at Digium that have told me that IAX2 was a work in progress but not ready for large scale prime time (isn't everything) and also that Realtime was also sub-par and needed to be re-written from scratch. This was a couple of years ago. <br>
<br>Thanks,<br>Steve Totaro<br></div>