[Asterisk-Users] codec issues

[Shoval Tomer]

Thanks Steve,
See my answers inline

> -----Original Message-----
> From: Steve Kann [mailto:stevek at stevek.com]
> Sent: Tuesday, December 21, 2004 1:49 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] codec issues
> 
> Shoval Tomer wrote:
> 
> >We've bought the G729 codec for lowering SIP bandwidth usage (we're
> >using grandstream phones) and we're quite happy with it up until I
tried
> >using IAXPhone 0.2.0 build 116 with my asterisk 1.0.0 installations.
> >
> >Weirdly enough, calls from IAXphone to the GS phone work just fine.
> >So are calls from both phones to voicemail. And from both phones to
> >analog phones connected to FXS ports.
> >
> >Calls from GS to IAXphone ring, and once I answer the call in
IAXphone,
> >I hear a very load noise.
> >
> >
> On which side do you hear this noise? IAXphone, or GS?
> 

I hear the noise on IAX site

> Does the noise continue, or just go for a few milliseconds?

Continue till I hang up

> 
> Does your version of IAXphone support multiple codecs, or just GSM?


Supports only GSM

> 
> >Asterisk CLI shows this:
> >channel.c:1314 ast_read: Dropping incompatible voice frame on
IAX2/205/1
> >of format GSM since our native format has changed to G729A
> >
> >(not just once, over and over and over again till I hang up)
> >
> >my sip.conf entry for the grandstream phone shows
> >disallow=all
> >allow=g729
> >and
> >reinvite=no
> >
> >I did 'iax2 show channels' and 'sip show channels'
> >
> >When I call from IAXPhone to GS, the IAX2 channel shows codec GSM and
> >the Sip channel shows codec G729A
> >
> >When I call the other way around, Sip shows G729A and IAX2 shows GSM.
> >
> >Hmm, seems ok...
> >
> >I tried changing my sip conf to include allow=g729,gsm
> >
> >Now the calls sounds fine, but the bandwidth is uses is near 20K
instead
> >of just 6K (both phones are near me, and the Asterisk server is at a
> >remote location, and I can monitor bandwidth usage in my FW).
> >
> >Can anyone help?
> >
> >
> 
> It obviosuly sounds like codec negotion, on one side or another, isn't
> working, and you're sending an incompatible codec to the other side,
or
> the other side doesn't know what codec is being sent..
>

How can I even control that?

And sip show channels and iax2 show channels show the correct codecs.
 
> Using ethereal to see what's happening on the network would show you
> what's going on pretty clearly..
> 
> -SteveK
> 

I'll try but it's not going to be easily done, as the asterisk server is
at a remote location and I'm no ethereal expert...

> 
> 
> 
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