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<p class=MsoNormal><span style='font-size:9.0pt;font-family:"Trebuchet MS","sans-serif";
color:navy'>Very good explanation.<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:9.0pt;font-family:"Trebuchet MS","sans-serif";
color:navy'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='font-size:9.0pt;font-family:"Trebuchet MS","sans-serif";
color:navy'>Thanks!<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:9.0pt;font-family:"Trebuchet MS","sans-serif";
color:navy'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='font-size:9.0pt;font-family:"Trebuchet MS","sans-serif";
color:navy'>Aaron.<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:9.0pt;font-family:"Trebuchet MS","sans-serif";
color:navy'><o:p> </o:p></span></p>
<div style='border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0cm 0cm 0cm'>
<p class=MsoNormal><b><span lang=EN-US style='font-size:10.0pt;font-family:
"Tahoma","sans-serif"'>From:</span></b><span lang=EN-US style='font-size:10.0pt;
font-family:"Tahoma","sans-serif"'> admin@danofive.id.au
[mailto:admin@danofive.id.au] <b>On Behalf Of </b>Daniel<br>
<b>Sent:</b> Friday, 11 December 2009 12:30 PM<br>
<b>To:</b> Aaron Riemer<br>
<b>Cc:</b> cisco-voip@puck.nether.net<br>
<b>Subject:</b> Re: [cisco-voip] When is Transcoding Required?<o:p></o:p></span></p>
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<p class=MsoNormal><o:p> </o:p></p>
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<p class=MsoNormal> <o:p></o:p></p>
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<p class=MsoNormal>Yes you can selectively configure what codec is used between
phones, gateways and sites etc.... You can do this by using regions, also on
H323 dial peers you can set the codec.<o:p></o:p></p>
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<p class=MsoNormal> <o:p></o:p></p>
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<p class=MsoNormal>Regions<o:p></o:p></p>
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<p class=MsoNormal>Codecs can be configured between regions so that branch
talking to core over the WAN is a a G729 call between regions, you would set
the service parameter so that G729 is default between regions (inter region)
and then default that its G711 within a region (Intra region). You can then
manually configure specific regions, for example we use G711 to a Music on
Hold region so that we can do G711 MoH from a router flash and also in our
environment we have G711 configured to a tandberg trunk so that no matter what
site your at desktop video calls to our tandberg environment are G711
audio with a specific amount of video quality.<o:p></o:p></p>
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<p class=MsoNormal> <o:p></o:p></p>
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<p class=MsoNormal>If the gateway and phones are on the same LAN then use G711
have them in the same region. If the call is from a gateway to another site
phone on a different LAN over a WAN then use G729 between the regions.<o:p></o:p></p>
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<p class=MsoNormal> <o:p></o:p></p>
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<p class=MsoNormal>My example for this would be a site gateway receives an ISDN
call for a phone, If the call is answered at this point it would be G711, but
the user is not available so the call diverts to voicemail, the voicemail
server is centrallised at the head office so the call traverses the WAN to
conenct to the voicemail box, this is now between regions so the call is now
G729. The gateway dial peer would need to support G729, the voicemail server
would need to support G729. If they don't both support G729 then a
transcoder is needed your preference here to reduce bandwidth needed would be
to have the gateway support G729 via the dial peer and then a trancoder at head
office next to the voicemail server so that the traffic over the WAN from the
gateway is G729 to the transcoder and G711 to the G711 only voicemail server.<o:p></o:p></p>
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<p class=MsoNormal> <o:p></o:p></p>
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<p class=MsoNormal> <o:p></o:p></p>
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<p class=MsoNormal> <o:p></o:p></p>
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<p class=MsoNormal> <o:p></o:p></p>
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<p class=MsoNormal><br>
<br>
<o:p></o:p></p>
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<p class=MsoNormal>On Fri, Dec 11, 2009 at 1:49 PM, Aaron Riemer <<a
href="mailto:ariemer@wesenergy.com.au">ariemer@wesenergy.com.au</a>> wrote:<o:p></o:p></p>
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<div>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span
style='font-size:9.0pt;color:navy'>Thanks Daniel. So to make sure I understand
this correctly. Can Cisco IP phones dynamically use different codecs based on
where calls are going or coming from? If calls are coming from the PSTN then
would it make sense that the phone uses G.711 to reduce the level of
transcoding required? If calls are coming from the IP network then G.729 is
used? Can the call agent make this distinction of codec choice?</span><o:p></o:p></p>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span
style='font-size:9.0pt;color:navy'> </span><o:p></o:p></p>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span
style='font-size:9.0pt;color:navy'>As you can no doubt probably guess I am very
new to the area of voice and it is difficult to know where to start!</span><o:p></o:p></p>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span
style='font-size:9.0pt;color:navy'> </span><o:p></o:p></p>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span
style='font-size:9.0pt;color:navy'>Thanks, </span><o:p></o:p></p>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span
style='font-size:9.0pt;color:navy'> </span><o:p></o:p></p>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span
style='font-size:9.0pt;color:navy'>Aaron.</span><o:p></o:p></p>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><b><span
style='font-size:8.0pt;color:#339966'> </span></b><o:p></o:p></p>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span
style='font-size:9.0pt;color:navy'> </span><o:p></o:p></p>
<div style='border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0cm 0cm 0cm'>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><b><span
lang=EN-US style='font-size:10.0pt'>From:</span></b><span lang=EN-US
style='font-size:10.0pt'> <a href="mailto:admin@danofive.id.au" target="_blank">admin@danofive.id.au</a>
[mailto:<a href="mailto:admin@danofive.id.au" target="_blank">admin@danofive.id.au</a>]
<b>On Behalf Of </b>Daniel<br>
<b>Sent:</b> Friday, 11 December 2009 11:32 AM<br>
<b>To:</b> Aaron Riemer<br>
<b>Cc:</b> <a href="mailto:cisco-voip@puck.nether.net" target="_blank">cisco-voip@puck.nether.net</a>
<o:p></o:p></span></p>
<div>
<p class=MsoNormal><span lang=EN-US style='font-size:10.0pt'><br>
<b>Subject:</b> Re: [cisco-voip] When is Transcoding Required?<o:p></o:p></span></p>
</div>
</div>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'> <o:p></o:p></p>
<div>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>Hi
Aaron,<o:p></o:p></p>
</div>
<div>
<div>
<div>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'> <br>
If you have a site with an ISDN 10 with a LAN that has phones configured in the
same region as the gateway set to use G729, and the gateway voice dial peers
(H323) set to use G729 then you won't need transcoding as such. You
will still need DSP's for the packetisation from ISDN to IP packets. If
you use G729 then you'll require a PVDM2-32 as you'll use twice the amount
because of the codec complexity. If you used G711 from the gateway to phones
then you'll only need a PVDM2-16.<o:p></o:p></p>
</div>
<div>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'> <o:p></o:p></p>
</div>
<div>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>To
answer your question if a call comes in from the gateway requesting only
G711 to a device that is only G729 then yes a transcoder is required preferably
at the same LAN as the gateway and phone. For instance if a site gateway
recevies a call for a G711 UCCX call and the UCCX server is in another region
that requires the gateway to use G729 to UCCX then a transcoder is required to
transcode the G729 call to G711 so that the UCCX prompts can be heard.<o:p></o:p></p>
</div>
<div>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'> <o:p></o:p></p>
</div>
<div>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>This
would be the same for software conferencing from a branch site to the core
server, the branch phone and gateway would talk G729 to a transcoder which
would talk G711 to the software conference bridge. In this situation a hardware
conference bridge would be better utilised.<o:p></o:p></p>
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<div>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'> <o:p></o:p></p>
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<div>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>Use
this link below about number of calls per DSP etc... High Complexity codec
would mean G729 to name one.<o:p></o:p></p>
</div>
<div>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><a
href="http://www.cisco.com/en/US/prod/collateral/modules/ps3115/ps6024/prod_qas0900aecd8016c6ad_ps3115_Products_Q_and_A_Item.html"
target="_blank">http://www.cisco.com/en/US/prod/collateral/modules/ps3115/ps6024/prod_qas0900aecd8016c6ad_ps3115_Products_Q_and_A_Item.html</a><o:p></o:p></p>
</div>
<div>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'> <o:p></o:p></p>
</div>
<div>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>hope
that helps.<o:p></o:p></p>
</div>
<div>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'> <o:p></o:p></p>
</div>
<div>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>cheers,<o:p></o:p></p>
</div>
<div>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'> <o:p></o:p></p>
</div>
<div>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>Daniel<o:p></o:p></p>
</div>
<div>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'> <o:p></o:p></p>
</div>
<div>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><br>
<o:p></o:p></p>
</div>
<div>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>On
Fri, Dec 11, 2009 at 12:36 PM, Aaron Riemer <<a
href="mailto:ariemer@wesenergy.com.au" target="_blank">ariemer@wesenergy.com.au</a>>
wrote:<o:p></o:p></p>
<div>
<div>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span
style='font-size:9.0pt;color:navy'>Hi Guys,</span><o:p></o:p></p>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span
style='font-size:9.0pt;color:navy'> </span><o:p></o:p></p>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span
style='font-size:9.0pt;color:navy'>Can someone clarify exactly when transcoding
between codec’s will actually occur on voice gateways? For example if I
have a branch site with Cisco IP Telephony and the phones are using G.729 and
the site has a voice gateway and PSTN services in what situation will
transcoding occur? Will all inbound and outbound calls through the PSTN require
transcoding? i.e. G.729 to G.711 and vice versa? If the site has 10 lines will
I require at least 10 transcoding sessions to cater for 10 simultaneous PSTN
calls?</span><o:p></o:p></p>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span
style='font-size:9.0pt;color:navy'> </span><o:p></o:p></p>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span
style='font-size:9.0pt;color:navy'>Thanks,</span><o:p></o:p></p>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span
style='font-size:9.0pt;color:navy'> </span><o:p></o:p></p>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span
style='font-size:9.0pt;color:navy'>Aaron.</span><o:p></o:p></p>
</div>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><br>
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