[Asterisk-Users] use of SIP SHOW CHANNELS question

CW_ASN cw_asn at fibertel.com.ar
Sun Oct 19 21:11:40 MST 2003


I think the system calculates the cost in CPU milliseconds to translate each
codec with another one. In different machines (slower or faster), the costs
vary.
The machine that have G.729 codec loaded is a Pentium III 733 MHz, 128MB.

I assume this is a way to know which codecs was loaded, because if I unload
G.729 codec disappears from 'show translation' printout. But, who knows...

Regards,

Gus


----- Original Message -----
From: "Andrew Joakimsen" <andrew at envisionstudio.net>
To: <asterisk-users at lists.digium.com>
Sent: Monday, October 20, 2003 12:59 AM
Subject: RE: [Asterisk-Users] use of SIP SHOW CHANNELS question


> Why is mine different?
>
> localhost*CLI> show translation
>          Translation times between formats (in milliseconds)
>           Source Format (Rows) Destination Format(Columns)
>
>            G723    GSM   ULAW   ALAW  ADPCM  SLINR  LPC10  G729A  SPEEX
> ILBC
>     G723      -     45     41     41     41     40     46      -      -
> 73
>      GSM    713      -      2      2      2      1      7      -      -
> 34
>     ULAW    713      6      -      1      2      1      7      -      -
> 34
>     ALAW    713      6      1      -      2      1      7      -      -
> 34
>      MP3    723     16     12     12     12     11     17      -      -
> 44
>    ADPCM    713      6      2      2      -      1      7      -      -
> 34
>    SLINR    712      5      1      1      1      -      6      -      -
> 33
>    LPC10    717     10      6      6      6      5      -      -      -
> 38
>    G729A      -      -      -      -      -      -      -      -      -
> -
>    SPEEX      -      -      -      -      -      -      -      -      -
> -
>     ILBC    718     11      7      7      7      6     12      -      -
> -
>
> Notice the G723 almost 4x higher.
>
>
> > -----Original Message-----
> > From: asterisk-users-admin at lists.digium.com [mailto:asterisk-users-
> > admin at lists.digium.com] On Behalf Of CW_ASN
> > Sent: Sunday, October 19, 2003 11:43 PM
> > To: asterisk-users at lists.digium.com
> > Subject: Re: [Asterisk-Users] use of SIP SHOW CHANNELS question
> >
> > pamAssassin 2.55 (1.174.2.19-2003-05-19-exp)
> >
> > How about 'show translations' command?
> >
> > This machine doesn't have G.729 codec licenses...
> > AFAIK, this command calculates the cost for each translation...
> >
> > noc2pbx*CLI> show translation
> >          Translation times between formats (in milliseconds)
> >           Source Format (Rows) Destination Format(Columns)
> >
> >            G723    GSM   ULAW   ALAW  ADPCM  SLINR  LPC10  G729A
> SPEEX
> > ILBC
> >
> > 23      -      -      -      -      -      -      -      -      -
> -
> >      GSM      -      -      2      2      2      1     13      -
> -
> > 38
> >     ULAW      -      6      -      1      2      1     13      -
> -
> > 38
> >     ALAW      -      6      1      -      2      1     13      -
> -
> > 38
> >      MP3      -     15     11     11     11     10     22      -
> -
> > 47
> >    ADPCM      -      6      2      2      -      1     13      -
> -
> > 38
> >    SLINR      -      5      1      1      1      -     12      -
> -
> > 37
> >    LPC10      -      9      5      5      5      4      -      -
> -
> > 41
> >
> > 9A      -      -      -      -      -      -      -      -      -
> -
> >
> > EX      -      -      -      -      -      -      -      -      -
> -
> >     ILBC      -     11      7      7      7      6
> > 8      -      -      -
> > noc2pbx*CLI>
> >
> > And this machine has G.729 loaded:
> >
> > noc2pbx2*CLI> show translation
> >          Translation times between formats (in milliseconds)
> >           Source Format (Rows) Destination Format(Columns)
> >
> >            G723    GSM   ULAW   ALAW  ADPCM  SLINR  LPC10  G729A
> SPEEX
> > ILBC
> >     G723      -     62     56     56     56     55     64    106
> -
> > 106
> >      GSM    240      -      3      3      3      2     11     53
> -
> > 53
> >     ULAW    239      8      -      1      2      1     10     52
> -
> > 52
> >     ALAW    239      8      1      -      2      1     10     52
> -
> > 52
> >      MP3    258     27     21     21     21     20     29     71
> -
> > 71
> >    ADPCM    239      8      2      2      -      1     10     52
> -
> > 52
> >    SLINR    238      7      1      1      1      -      9     51
> -
> > 51
> >    LPC10    245     14      8      8      8      7      -     58
> -
> > 58
> >    G729A 100237 100006 100000 100000 100000  99999 100008      -
> -
> > 100050
> >
> > EX      -      -      -      -      -      -      -      -      -
> -
> >     ILBC    247     16     10     10     10      9     18
> > 0      -      -
> > noc2pbx2*CLI>
> >
> >
> >
> > Regards,
> >
> > Gus
> >
> > ----- Original Message -----
> > From: "Tilghman Lesher" <tilghman at mail.jeffandtilghman.com>
> > To: <asterisk-users at lists.digium.com>
> > Sent: Sunday, October 19, 2003 11:44 PM
> > Subject: Re: [Asterisk-Users] use of SIP SHOW CHANNELS question
> >
> >
> > > On Sunday 19 October 2003 21:30, Andrew Kohlsmith wrote:
> > > > > *CLI> show codec 4
> > > > > No such command 'show codec' (type 'help' for help)
> > > > > *CLI> show audio codecs
> > > > > No such command 'show audio' (type 'help' for help)
> > > >
> > > > Rebuild your system.  I had that happen and my Makefile was
> screwed
> > > > up and didn't actually build any codecs.
> > >
> > > Stop already.  The command serves only as a translation table.  It
> does
> > > NOT, repeat NOT, repeat NOT, suggest anything about which codecs
> > > are actually loaded.
> > >
> > > If you actually run this now, you should get the following message
> on
> > > your console (put there SPECIFICALLY because people were confusing
> > > what the command actually does):
> > >
> > > Disclaimer: this command is for informational purposes only.
> > >         It does not indicate anything about your configuration.
> > >
> > > -Tilghman
> > >
> > > _______________________________________________
> > > 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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