[Asterisk-Users] G723 (was SIP using which codec?)

[Eric Wieling]

If you want to be able to use G723 from a legal standpoint you will have
to license the codec from the current patent holders.  The patent
holder's price list can be found at
http://www.dspg.com/technology/LicensePricing.html

If you obtain a license to use G723 then Digium or the Asterisk user
community might be able to assist you in adding the codec to Asterisk. 
I don't know how this would work from the technical standpoint.

For G729, Voiceage (the patent holder for G729) provides a binary module
with some horrible license agreement to add to Asterisk.  You can buy
G729 licenses direct from Digium for $10/channel.

If you can't afford the US$30,000+ licensing fees for G723 and if you
can't use G729 or any other codec then you really are out of luck.

There are many people that want to use G723 (myself included), but I'm
not going to spend that kind of money for four G729 channels.  I'd be
happy to pay $10/channel just like I have for the G729 license.

On Wed, 2003-08-20 at 12:28, Andrew Joakimsen wrote:
> And if one cannot use a different codec?
> 
> 
> -----Original Message-----
> From: asterisk-users-admin at lists.digium.com
> [mailto:asterisk-users-admin at lists.digium.com] On Behalf Of Eric Wieling
> Sent: Wednesday, August 20, 2003 9:51 AM
> To: asterisk-users at lists.digium.com
> Subject: Re: [Asterisk-Users] G723 (was SIP using which codec?)
> 
> MOH requires that Asterisk transcodes (It also has to transcode to for
> PSTN calls and voicemail and playing any sound files).  Asterisk can't
> transcode to or from G723.  Nope.  Doesn't work.  May very well never
> work.  Use a different codec.
> 
> On Wed, 2003-08-20 at 03:47, Andrew Joakimsen wrote:
> > Actually I got it working right before I gave up (I had the wrong line
> > in my config commented out)
> > 
> > But now I get these messages when I try to playback a recording:
> > 
> > NOTICE[16401]: File channel.c, Line 1406 (ast_set_write_format):
> Unable
> > to find a path from GSM to G723
> > WARNING[16401]: File file.c, Line 722 (ast_streamfile): Unable to open
> > transfer (format G723): No such file or directory
> > WARNING[16401]: File app_playback.c, Line 83 (playback_exec):
> > ast_streamfile failed on SIP/packet8.net-dab9 for transfer
> > 
> > And when I try to play music on hold:
> > 
> > NOTICE[16401]: File channel.c, Line 1406 (ast_set_write_format):
> Unable
> > to find a path from SLINR to G723
> > WARNING[16401]: File res_musiconhold.c, Line 421 (moh_alloc): Unable
> to
> > set 'SIP/packet8.net-dab9' to signed linear format
> > 
> > 
> > This is the missing link in my system, I greatly appreciate any help
> > that can be provided.
> > 
> > -----Original Message-----
> > From: asterisk-users-admin at lists.digium.com
> > [mailto:asterisk-users-admin at lists.digium.com] On Behalf Of John Todd
> > Sent: Wednesday, August 20, 2003 4:11 AM
> > To: asterisk-users at lists.digium.com
> > Subject: Re: [Asterisk-Users] SIP using which codec?
> > 
> > At 7:27 AM +0000 8/20/03, WipeOut . wrote:
> > >
> > >>  Is there a way to determine what codec the remote server wants to
> > use in
> > >>  a SIP session for an incoming call by looking at something,
> possiby
> > sip
> > >>  debug?
> > >>
> > >
> > >Take a look in the archives this was covered a couple of days ago..
> > >
> > >the command you are looking for is "sip show channels".. and then 
> > >look in the format column.. the formula for determining the format 
> > >was posted in the previous discussion and i can't rememebr it off 
> > >the top of my head..
> > >
> > >Later..
> > >
> > 
> > I think that the question is a bit more subtle than that.  The 
> > question says "wants to use", not "does use."
> > 
> > Currently, I think the only way you'll find this is with a SIP debug, 
> > looking at the SDP request.
> > 
> > JT
> > _______________________________________________
> > Asterisk-Users mailing list
> > Asterisk-Users at lists.digium.com
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> > 
> > _______________________________________________
> > Asterisk-Users mailing list
> > Asterisk-Users at lists.digium.com
> > http://lists.digium.com/mailman/listinfo/asterisk-users
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