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tony at softins.co.uk Guest
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Posted: Thu Feb 20, 2014 10:40 am Post subject: [asterisk-users] G729 - what happens if licences used up? |
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I haven't been able to find the answer online, and am not currently
able to conduct an experiment to find the answer...
I understand that in a SIP call where G729 has been negotiated as the
preferred codec, a G.729 licence is not consumed until there is a need
to perform transcoding, e.g. play a non-g729 sound, or do voicemail,
or enter a Meetme, etc.
What happens when a SIP call in progress needs a G.729 licence and
they are all in use already? Does the call fail, or go silent, or do a
re-INVITE to negotiate another codec?
I'm interested in what happens on Asterisk 1.2 (for a legacy system),
and also whether it is any different on later versions.
Thanks,
Tony
--
Tony Mountifield
Work: tony@softins.co.uk - http://www.softins.co.uk
Play: tony@mountifield.org - http://tony.mountifield.org
--
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paul.belanger at polyb... Guest
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Posted: Thu Feb 20, 2014 12:17 pm Post subject: [asterisk-users] G729 - what happens if licences used up? |
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On Thu, Feb 20, 2014 at 10:40 AM, Tony Mountifield <tony@softins.co.uk> wrote:
Quote: | I haven't been able to find the answer online, and am not currently
able to conduct an experiment to find the answer...
I understand that in a SIP call where G729 has been negotiated as the
preferred codec, a G.729 licence is not consumed until there is a need
to perform transcoding, e.g. play a non-g729 sound, or do voicemail,
or enter a Meetme, etc.
What happens when a SIP call in progress needs a G.729 licence and
they are all in use already? Does the call fail, or go silent, or do a
re-INVITE to negotiate another codec?
I'm interested in what happens on Asterisk 1.2 (for a legacy system),
and also whether it is any different on later versions.
| The question depends if you are offering up other codecs or not. If
you only using g729, the call will fail to establish because lack of
codecs. If you offer a both g729 and ulaw, then ulaw will be used.
--
Paul Belanger | PolyBeacon, Inc.
Jabber: paul.belanger@polybeacon.com | IRC: pabelanger (Freenode)
Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger
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EWieling at nyigc.com Guest
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Posted: Thu Feb 20, 2014 12:20 pm Post subject: [asterisk-users] G729 - what happens if licences used up? |
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In my experience when you run out of g729 licenses additional calls will fail. Simple as that. Make sure you run out of licenses.
-----Original Message-----
From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Tony Mountifield
Sent: Thursday, February 20, 2014 10:40 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] G729 - what happens if licences used up?
I haven't been able to find the answer online, and am not currently able to conduct an experiment to find the answer...
I understand that in a SIP call where G729 has been negotiated as the preferred codec, a G.729 licence is not consumed until there is a need to perform transcoding, e.g. play a non-g729 sound, or do voicemail, or enter a Meetme, etc.
What happens when a SIP call in progress needs a G.729 licence and they are all in use already? Does the call fail, or go silent, or do a re-INVITE to negotiate another codec?
I'm interested in what happens on Asterisk 1.2 (for a legacy system), and also whether it is any different on later versions.
Thanks,
Tony
--
Tony Mountifield
Work: tony@softins.co.uk - http://www.softins.co.uk
Play: tony@mountifield.org - http://tony.mountifield.org
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
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tony at softins.co.uk Guest
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Posted: Thu Feb 20, 2014 12:28 pm Post subject: [asterisk-users] G729 - what happens if licences used up? |
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In article <CALLKq0RpimD05jz=OsBGjyDx-41UeBOhXMFT_SKwFJT51kohwQ@mail.gmail.com>,
Paul Belanger <paul.belanger@polybeacon.com> wrote:
Quote: | On Thu, Feb 20, 2014 at 10:40 AM, Tony Mountifield <tony@softins.co.uk> wrote:
Quote: | I haven't been able to find the answer online, and am not currently
able to conduct an experiment to find the answer...
I understand that in a SIP call where G729 has been negotiated as the
preferred codec, a G.729 licence is not consumed until there is a need
to perform transcoding, e.g. play a non-g729 sound, or do voicemail,
or enter a Meetme, etc.
What happens when a SIP call in progress needs a G.729 licence and
they are all in use already? Does the call fail, or go silent, or do a
re-INVITE to negotiate another codec?
I'm interested in what happens on Asterisk 1.2 (for a legacy system),
and also whether it is any different on later versions.
| The question depends if you are offering up other codecs or not. If
you only using g729, the call will fail to establish because lack of
codecs. If you offer a both g729 and ulaw, then ulaw will be used.
|
The codecs offered by each end would be g729, alaw and ulaw.
I guess my point is that the licence is NOT required to negotiate codecs
and establish the call, e.g. if g.729 sounds are installed and calls are
pass-through, then no transcoding is required.
So the call will negotiate g729 and get established, and then if later
the dialplan calls something that requires transcoding, the licence is
requested at that time. What happens if there is not one available?
Can/will it do a re-INVITE to change codec, or does the call fail,
or does it continue but go silent?
Cheers
Tony
--
Tony Mountifield
Work: tony@softins.co.uk - http://www.softins.co.uk
Play: tony@mountifield.org - http://tony.mountifield.org
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users |
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mailinglist+asterisk a... Guest
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Posted: Thu Feb 20, 2014 12:33 pm Post subject: [asterisk-users] G729 - what happens if licences used up? |
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On 20/02/14 17:16, Paul Belanger wrote:
Quote: | On Thu, Feb 20, 2014 at 10:40 AM, Tony Mountifield<tony@softins.co.uk> wrote:
Quote: | I haven't been able to find the answer online, and am not currently
able to conduct an experiment to find the answer...
I understand that in a SIP call where G729 has been negotiated as the
preferred codec, a G.729 licence is not consumed until there is a need
to perform transcoding, e.g. play a non-g729 sound, or do voicemail,
or enter a Meetme, etc.
What happens when a SIP call in progress needs a G.729 licence and
they are all in use already? Does the call fail, or go silent, or do a
re-INVITE to negotiate another codec?
I'm interested in what happens on Asterisk 1.2 (for a legacy system),
and also whether it is any different on later versions.
| The question depends if you are offering up other codecs or not. If
you only using g729, the call will fail to establish because lack of
codecs. If you offer a both g729 and ulaw, then ulaw will be used.
| That would only apply for new calls. Even new calls would still
typically accept g729 even if there are no licenses remaining as there
might not be transcoding required.
What I would expect to happen if there were no licenses is for you to
see an error on the console (possibly repeated multiple times) and for
there to be no audio. This is certainly what happens if you have a g729
call with no license and then try to play a sound file which does not
have a native g729 format.
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
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