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

Ross Finlayson finlayson at live.com
Tue Mar 11 19:48:45 PST 2003

FYI, the Asterisk software PBX <http://www.asterisk.org/> has now 
incorporated my recent patches to support dynamic RTP payload types.  As a 
consequence, its SIP implementation now supports Speex, so if you have a 
Speex-compatible SIP client, you can use it to make calls using Asterisk.

Some caveats:
- 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.)
- Each outgoing RTP packet (from Asterisk) contains just a single Speex 
frame.  Similarly, each incoming RTP packet (from a client) should contain 
just a single Speex frame.
- 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.

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.)


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