asterisk users - May 2010 - LAN IAX2 trunk bad audio quality vs. LAN SIP trunk good audio quality

Hi,

I have an audio quality problem regarding IAX2. I have 2 Asterisk servers
interconnected via 2 LAN trunks at 1Gbps (no nat, no firewall).
One trunk is SIP and the other IAX2.
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).

So lately I setup calls to go through the SIP trunk and audio quality is OK (no
"third overlapping noise").

This is happening between Asterisk 1.4.31 and a 1.2.40.

I'm wondering if there's something I can tweak in IAX2 to eliminate this
artifact.

Could the IAX2 jitter buffer between 1.2 and 1.4 be an issue (I believe it's
enabled by default)?

Thanks,

Vieri

Show the details on the active channels when using both methods and 
check what codecs are being used.

Vieri wrote:
> Hi,
> 
> I have an audio quality problem regarding IAX2. I have 2 Asterisk servers
interconnected via 2 LAN trunks at 1Gbps (no nat, no firewall).
> One trunk is SIP and the other IAX2.
> 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).
> 
> So lately I setup calls to go through the SIP trunk and audio quality is OK
(no "third overlapping noise").
> 
> This is happening between Asterisk 1.4.31 and a 1.2.40.
> 
> I'm wondering if there's something I can tweak in IAX2 to eliminate
this artifact.
> 
> Could the IAX2 jitter buffer between 1.2 and 1.4 be an issue (I believe
it's enabled by default)?
> 
> Thanks,
> 
> Vieri
> 
> 
> 
>       
>

On Thu, May 13, 2010 at 4:17 AM, Vieri <rentorbuy at yahoo.com> wrote:
> Hi,
>
> I have an audio quality problem regarding IAX2. I have 2 Asterisk servers
> interconnected via 2 LAN trunks at 1Gbps (no nat, no firewall).
> One trunk is SIP and the other IAX2.
> 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).
>
> So lately I setup calls to go through the SIP trunk and audio quality is OK
> (no "third overlapping noise").
>
> This is happening between Asterisk 1.4.31 and a 1.2.40.
>
> I'm wondering if there's something I can tweak in IAX2 to eliminate
this
> artifact.
>
> Could the IAX2 jitter buffer between 1.2 and 1.4 be an issue (I believe
> it's enabled by default)?
>
> Thanks,
>
> Vieri
>
>
> 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"

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.

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.

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.

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.

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.

Thanks,
Steve Totaro
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