asterisk users - Feb 2006 - Codec order sent wrong from Asterisk

I'm communicating a softphone (SJPhone) to a Grandstream phone GXP-2000.
The codec order on each one is the next:

SJPhone: GSM - iLBC - PCMA - PCMU
GXP2000: G729 - GSM - PCMA - PCMU

(I have a G729 license, so there's no problem with transcoding G729)

In my sip.conf, I've defined the following codec order:

disallow=all
allow=g729
allow=gsm
allow=g726
allow=alaw
allow=ulaw

And my peers shows this order correctly:

Codecs       : 0x11e (gsm|ulaw|alaw|g726|g729)
Codec Order  : (g729,gsm,g726,alaw,ulaw)

Canreinvite is set to NO.

But, if I initiate a call from the softphone to GXP-2000, Asterisk
always to the GXP phone GSM as the first codec choice, instead of G729,
as I could check with ethereal running in the same server than Asterisk.
The SIP INVITE from Asterisk to GXP looks like

*****************************************************************
Request-Line: INVITE sip:5805907@192.168.1.105;user=phone SIP/2.0
Message Header
Message body
     Session Description Protocol
         Session Description Protocol Version (v): 0
         Owner/Creator, Session Id (o): root 27682 27682 IN IP4 192.168.1.2
         Session Name (s): session
         Connection Information (c): IN IP4 192.168.1.2
         Time Description, active time (t): 0 0
         Media Description, name and address (m): audio 14224 RTP/AVP 3 
18 111 8 0
             Media Type: audio
             Media Port: 14224
             Media Proto: RTP/AVP
             Media Format: GSM 06.10
             Media Format: ITU-T G.729
             Media Format: 111
             Media Format: ITU-T G.711 PCMA
             Media Format: ITU-T G.711 PCMU
         Media Attribute (a): rtpmap:3 GSM/8000
             Media Attribute Fieldname: rtpmap
             Media Attribute Value: 3 GSM/8000
         Media Attribute (a): rtpmap:18 G729/8000
             Media Attribute Fieldname: rtpmap
             Media Attribute Value: 18 G729/8000
         Media Attribute (a): fmtp:18 annexb=no
             Media Attribute Fieldname: fmtp
             Media Attribute Value: 18 annexb=no
         Media Attribute (a): rtpmap:111 G726-32/8000
             Media Attribute Fieldname: rtpmap
             Media Attribute Value: 111 G726-32/8000
         Media Attribute (a): rtpmap:8 PCMA/8000
             Media Attribute Fieldname: rtpmap
             Media Attribute Value: 8 PCMA/8000
         Media Attribute (a): rtpmap:0 PCMU/8000
             Media Attribute Fieldname: rtpmap
             Media Attribute Value: 0 PCMU/8000
         Media Attribute (a): silenceSupp:off - - - -
             Media Attribute Fieldname: silenceSupp
             Media Attribute Value: off - - - -
*****************************************************************

So it can be clearly seen how GSM is before G729.

Anybody knows if this is an existing bug? Or am I doing something wrong?
Thanks a lot for your attention.

-- 
Atly.
Alvaro Palma

?lvaro Palma wrote:
> But, if I initiate a call from the softphone to GXP-2000, Asterisk
> always to the GXP phone GSM as the first codec choice, instead of G729,
> as I could check with ethereal running in the same server than Asterisk.
> The SIP INVITE from Asterisk to GXP looks like
This is intentional, and well documented on the wiki. Asterisk attempts
to use the same codec for the outbound call as the inbound call (if it
is allowed), to reduce the need for transcoding.

hi Palma,
as the SJ initiate the call, it will allways go with GSM Codec as the codec
should be identical used on both sides. as you do not have G729 on the SJ,
it will never use G729.
furthermore, i think that if the GSM will not work, then the second option
choosed would be PCMA

i hope i gave you a way further.

Mickey


On 2/23/06, ?lvaro Palma <apalma@opschile.cl>
wrote:
>
> I'm communicating a softphone (SJPhone) to a Grandstream phone
GXP-2000.
> The codec order on each one is the next:
>
> SJPhone: GSM - iLBC - PCMA - PCMU
> GXP2000: G729 - GSM - PCMA - PCMU
>
> (I have a G729 license, so there's no problem with transcoding G729)
>
> In my sip.conf, I've defined the following codec order:
>
> disallow=all
> allow=g729
> allow=gsm
> allow=g726
> allow=alaw
> allow=ulaw
>
> And my peers shows this order correctly:
>
> Codecs       : 0x11e (gsm|ulaw|alaw|g726|g729)
> Codec Order  : (g729,gsm,g726,alaw,ulaw)
>
> Canreinvite is set to NO.
>
> But, if I initiate a call from the softphone to GXP-2000, Asterisk
> always to the GXP phone GSM as the first codec choice, instead of G729,
> as I could check with ethereal running in the same server than Asterisk.
> The SIP INVITE from Asterisk to GXP looks like
>
> *****************************************************************
> Request-Line: INVITE sip:5805907@192.168.1.105;user=phone SIP/2.0
> Message Header
> Message body
>     Session Description Protocol
>         Session Description Protocol Version (v): 0
>         Owner/Creator, Session Id (o): root 27682 27682 IN IP4 192.168.1.2
>         Session Name (s): session
>         Connection Information (c): IN IP4 192.168.1.2
>         Time Description, active time (t): 0 0
>         Media Description, name and address (m): audio 14224 RTP/AVP 3
> 18 111 8 0
>             Media Type: audio
>             Media Port: 14224
>             Media Proto: RTP/AVP
>             Media Format: GSM 06.10
>             Media Format: ITU-T G.729
>             Media Format: 111
>             Media Format: ITU-T G.711 PCMA
>             Media Format: ITU-T G.711 PCMU
>         Media Attribute (a): rtpmap:3 GSM/8000
>             Media Attribute Fieldname: rtpmap
>             Media Attribute Value: 3 GSM/8000
>         Media Attribute (a): rtpmap:18 G729/8000
>             Media Attribute Fieldname: rtpmap
>             Media Attribute Value: 18 G729/8000
>         Media Attribute (a): fmtp:18 annexb=no
>             Media Attribute Fieldname: fmtp
>             Media Attribute Value: 18 annexb=no
>         Media Attribute (a): rtpmap:111 G726-32/8000
>             Media Attribute Fieldname: rtpmap
>             Media Attribute Value: 111 G726-32/8000
>         Media Attribute (a): rtpmap:8 PCMA/8000
>             Media Attribute Fieldname: rtpmap
>             Media Attribute Value: 8 PCMA/8000
>         Media Attribute (a): rtpmap:0 PCMU/8000
>             Media Attribute Fieldname: rtpmap
>             Media Attribute Value: 0 PCMU/8000
>         Media Attribute (a): silenceSupp:off - - - -
>             Media Attribute Fieldname: silenceSupp
>             Media Attribute Value: off - - - -
> *****************************************************************
>
> So it can be clearly seen how GSM is before G729.
>
> Anybody knows if this is an existing bug? Or am I doing something wrong?
> Thanks a lot for your attention.
>
> --
> Atly.
> Alvaro Palma
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
>
> Asterisk-Users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
http://lists.digium.com/pipermail/asterisk-users/attachments/20060224/30012d72/attachment.htm

Can the asterisk support a "coaching function" for the Supervisor to
tap
onto a call and coach the agent as she speaks to the customer without 
the customer hearing it.?

Customer database management softward (or CRM) - is this being included?

Best regards
Stephen
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
http://lists.digium.com/pipermail/asterisk-users/attachments/20060224/71a7cdef/attachment.htm