[speex-dev] Speex SIP support in the "Asterisk" PBX, FYI

Ross Finlayson finlayson at live.com
Tue Mar 11 20:13:48 PST 2003

At 07:55 PM 3/11/03, Jean-Marc Valin wrote:
> > - Only narrowband (8 kHz) Speex is currently supported; not
> > wideband.  (Unfortunately, the assumption that audio sample rate == 8 kHz
> > is riddled throughout the Asterisk code.)
>Perhaps it's still possible to send wideband, while telling Asterisk
>it's narrowband (the bit-stream is such that you can decode a wideband
>frame even if you think it's narrowband).

Unfortunately, the sampling frequency is also the RTP timestamp frequency, 
so if Asterisk thinks that the stream is 8 kHz, but the Speex client thinks 
that it's 16 kHz, then each side will probably misinterpret the RTP 
timestamps sent by the other side.

> > Also, FYI, my command-line SIP client "playSIP"
> > <http://www.live.com/playSIP/> can also record - into a file - the Speex
> > audio from a SIP call.  (Unfortunately, this output file will be raw Speex
> > data, rather than an 'ogg' format file, so "speexdec" currently can't
> > decode it.)
>I see a potential problem here. Because Speex has multiple bit-rate, you
>can't just dump frames to a file and try to decode them after. You need
>to either use Ogg or at least encode the size of each packet so the
>decoder knows how long it needs to read.

I'll probably ending up adding an option to output the data as an 'ogg' 
format file (just as there's currently an option to output the data as a 
'.mov' file).

         Ross Finlayson

<p><p><p>>         Jean-Marc
>Jean-Marc Valin, M.Sc.A.
>LABORIUS (http://www.gel.usherb.ca/laborius)
>Université de Sherbrooke, Québec, Canada

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