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