[asterisk-dev] Re: Help with 240 samples on frames read from chan_iax

[Moises Silva]

Tony, thanks for the suggestion. Yes, I remembered an issue with ILBC,
but the phone on the other side is using ULAW as well. I tried to
avoid any transcoding to see if that helped, but so far, no look. Im
still looking the code to find a solution.

Any other ideas out there?

On 11/4/06, Tony Mountifield <tony at softins.clara.co.uk> wrote:
> In article <c4d05cbe0611041120q602b5ec2k6002589394cd30f6 at mail.gmail.com>,
> Moises Silva <moises.silva at gmail.com> wrote:
> > Hi everyone. I know app_conference is not formal part from Asterisk,
> > but I think you may help me a bit to understand what is happening
> > here.
> >
> >   app_conference is working just fine, except when receives frames
> > with 240 samples, it seems is somehow hardcoded to expect frames of
> > 160 samples, so when receives 240, a buffer overflow ocurrs on the mix
> > buffer and crashes Asterisk. Frames with 240 samples, so far, are just
> > generated by my IAX2 connection with other server, so, if I enter a
> > conference with a SIP channel using ULAW, ZAP channel using SLIN, and
> > IAX2 channel (kiax softphone) with ULAW, everything works fine, but at
> > the moment the other IAX2 server enters the conference (ULAW also),
> > everything crashes.
> >
> >   I can stop asterisk from crashing modifing the app_conference code
> > to NOT mix 240 samples frames, and works, but obviously the voice from
> > the IAX2 server is not received by the other parties.
> >
> > Can somebody just give me a hint about where to look?
> >
> > Why just that IAX2 server generates frames with 240 samples?
>
> One possible idea - is the phone on the other side of that server
> negotiating the iLBC codec with 30ms packet size?
>
> Cheers
> Tony
> --
> Tony Mountifield
> Work: tony at softins.co.uk - http://www.softins.co.uk
> Play: tony at mountifield.org - http://tony.mountifield.org
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
>
> asterisk-dev mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-dev
>


-- 
"Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org"