--- On Thu, 5/13/10, Gareth Blades <list-aster...@skycomuk.com> wrote:
> Show the details on the active
> channels when using both methods and
> check what codecs are being used.
The audio codecs are different:
Type: SIP
State: Up (6)
Rings: 0
NativeFormats: 0x4 (ulaw)
WriteFormat: 0x40 (slin)
ReadFormat: 0x40 (slin)
WriteTranscode: Yes
ReadTranscode: Yes
Type: IAX2
State: Up (6)
Rings: 0
NativeFormats: 0x8 (alaw)
WriteFormat: 0x8 (alaw)
ReadFormat: 0x8 (alaw)
WriteTranscode: No
ReadTranscode: No
By the way, I have this in iax.conf:
[interboxIAX2]
deny=all
allow=ulaw
allow=gsm
type=friend
host=192.168.250.111
secret=mysecret
auth=plaintext
requirecalltoken=no
qualify=yes
context=mycontext
trunk=yes
username=interbox
Shouldn't the channel details report ulaw instead of alaw?
Also, if I change [interboxIAX2] and replace ulaw with alaw, the result is the
same (I still experience bad audio quality).
Maybe I should try slin but how do I "force it"?
> 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
> >
> >
> >
> >
> >
>
>
> --
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