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

[Moises Silva]

Dan Austin:

Thanks for your suggestion, on one side of the link im using
Asterisk-1.2.1, in the app_conference side im using Asterisk-1.2.12.1.
But it seems that is not relevant. You were right, the Linksys SPA im
using sends 30ms RTP packetization,  using a grandstream phone made it
work.

Tony:
Thank you very much for your help too


   Well, now knowing the fundamental cause, im going to the hardest
part, finding a solution for the issue on app_conference code, I dont
think is a good idea rely on 20ms frames. Or is it possible to make
Asterisk use only 20ms frames?

Kind Regards

On 11/5/06, Tony Mountifield <tony at softins.clara.co.uk> wrote:
> In article <c4d05cbe0611041503r446cc0b2p3031f0e6de7fa7e7 at mail.gmail.com>,
> Moises Silva <moises.silva at gmail.com> wrote:
> > 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?
>
> A couple:
>
> 1) Does your iax.conf on the rogue IAX box have a setting for trunkfreq?
> If it was set at trunkfreq=30 (I suppose it should really be called
> trunkperiod, not trunkfreq), and trunking is enabled, that could explain
> the 240-sample frames.
>
> 2) Try running ethereal or tcpdump on that box to capture both the IAX
> stream and the stream from the phone on the other side. It's not the
> codec that is at issue with the remote phone, but rather the phone's
> setting for frame size. If it is sending 30ms frames, even in uLaw, it
> may be that they are getting relayed through at the same size.
>
> Hope this helps
>
> Cheers
> Tony
>
> > 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
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>
> --
> Tony Mountifield
> Work: tony at softins.co.uk - http://www.softins.co.uk
> Play: tony at mountifield.org - http://tony.mountifield.org
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