[Speex-dev] app_conference(Asterisk) with Speex

[Jean-Marc Valin]

> Asterisk expects speex frames to have a terminator.   

Then it might just be that Linphone is using a version of Speex that had
a bug in the terminator code. Jonathan, try making sure Linphone uses
Speex 1.1.11.1

> Also, since then, I have written, and added to asterisk and iaxclient (I 
> think I probably posted the code to this list, but if not, and if you're 
> interested in putting it in libspeex, that would be nice too), a couple 
> of functions which can quickly parse speex data to determine how many 
> 20ms frames they contain.
> 
> the function is: *static* *int* *speex_get_samples*(*unsigned* *char* 
> *data, *int* len), and you just point it at some speex data, it it 
> returns the number of samples that are there.  Look for it at: 
> http://cvs.sourceforge.net/viewcvs.py/iaxclient/iaxclient/lib/libiax2/src/iax.c?rev=1.71&view=markup

That would probably be useful. If you ever feel like implementing it in
terms of SpeexBits and internal Speex mode structs, I will add it to
Speex.

	Jean-Marc

> -SteveK
> 
> 
> 
> 
> 
> >	Jean-Marc
> >
> >Le mardi 31 janvier 2006 à 10:43 -0500, Steve Kann a écrit :
> >  
> >
> >>jonathan blais wrote: 
> >>    
> >>
> >>>I'm using Linphone. I tested with Asterisk and Speex only, I created
> >>>a channel with echo and it worked. It seems to have problem when
> >>>using app_conference.
> >>>      
> >>>
> >>If you just use app_echo, then asterisk won't be trying to decode your
> >>frames; it will just be sending them back to you.  Therefore, if your
> >>client is using an incompatible packing of the frames, it won't
> >>matter, as long as your client can also understand it.
> >>
> >>In order to determine if your client is using a compatible packing,
> >>you'll need to have asterisk do something which requires asterisk to
> >>decode the frames:  Like bridge the call to a channel using another
> >>codec, or record the call to a non-speex format (i.e. write to a PCM
> >>WAV file), etc.
> >>
> >>-SteveK
> >>
> >>    
> >>
> >>>Jonathan
> >>>
> >>>2006/1/31, Steve Kann <stevek at stevek.com>: 
> >>>        jonathan blais wrote:
> >>>        
> >>>        > Hi,
> >>>        >
> >>>        > Does anyone ever used Speex with app_conference in
> >>>        Asterisk ? I'm
> >>>        > having a hard time to figure why I always get this error
> >>>        "warning:
> >>>        > Invalid mode encountered: corrupted stream?". 
> >>>        
> >>>        Yes, we do this all the time.  Where are your speex frames
> >>>        coming from?
> >>>        There's some SIP phones out there which seem to use an
> >>>        incompatible
> >>>        speex encoding, I seem to recall.
> >>>        
> >>>        
> >>>        -SteveK
> >>>        
> >>>        
> >>>        
> >>>
> >>>      
> >>>
> >>_______________________________________________
> >>Speex-dev mailing list
> >>Speex-dev at xiph.org
> >>http://lists.xiph.org/mailman/listinfo/speex-dev
> >>    
> >>
> 
> 
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