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

Jean-Marc Valin jean-marc.valin at hermes.usherb.ca
Tue Mar 11 19:55:31 PST 2003

> - 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).

> - Some existing clients (such as "linphone") will need to be modified to 
> use the string "speex" rather than "speex-<version-number>" in their SDP 
> "a=rtpmap:" line.

The latest version of Linphone had been made up-to-date with the RTP
spec, so this is no longer a problem.

> 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.


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

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