Speex dev - Aug 2004 - Speex SIP support in the "Asterisk" PBX, FYI

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.)

        Ross.

--- >8 ----
List archives:  http://www.xiph.org/archives/
Ogg project homepage: http://www.xiph.org/ogg/
To unsubscribe from this list, send a message to
'speex-dev-request@xiph.org'
containing only the word 'unsubscribe' in the body.  No subject is
needed.
Unsubscribe messages sent to the list will be ignored/filtered.

> - 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


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

<p>
-------------- next part --------------
A non-text attachment was scrubbed...
Name: signature.asc
Type: application/pgp-signature
Size: 242 bytes
Desc: signature.asc
Url :
http://lists.xiph.org/pipermail/speex-dev/attachments/20030311/c8eab276/signature.pgp

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
         LIVE.COM
         <http://www.live.com/>

<p><p><p>>         Jean-Marc
>
>--
>Jean-Marc Valin, M.Sc.A.
>LABORIUS (http://www.gel.usherb.ca/laborius)
>Université de Sherbrooke, Québec, Canada
>
--- >8 ----
List archives:  http://www.xiph.org/archives/
Ogg project homepage: http://www.xiph.org/ogg/
To unsubscribe from this list, send a message to
'speex-dev-request@xiph.org'
containing only the word 'unsubscribe' in the body.  No subject is
needed.
Unsubscribe messages sent to the list will be ignored/filtered.