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<div>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.</div>
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<div>Regions</div>
<div>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.</div>
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<div>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.</div>
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<div>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.</div>
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<div class="gmail_quote">On Fri, Dec 11, 2009 at 1:49 PM, Aaron Riemer <span dir="ltr"><<a href="mailto:ariemer@wesenergy.com.au">ariemer@wesenergy.com.au</a>></span> wrote:<br>
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<p class="MsoNormal"><span style="FONT-SIZE: 9pt; 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></p>
<p class="MsoNormal"><span style="FONT-SIZE: 9pt; COLOR: navy"> </span></p>
<p class="MsoNormal"><span style="FONT-SIZE: 9pt; 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></p>
<p class="MsoNormal"><span style="FONT-SIZE: 9pt; COLOR: navy"> </span></p>
<p class="MsoNormal"><span style="FONT-SIZE: 9pt; COLOR: navy">Thanks, </span></p>
<p class="MsoNormal"><span style="FONT-SIZE: 9pt; COLOR: navy"> </span></p>
<p class="MsoNormal"><span style="FONT-SIZE: 9pt; COLOR: navy">Aaron.</span></p>
<p class="MsoNormal"><b><span style="FONT-SIZE: 8pt; COLOR: #339966"> </span></b></p>
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<p class="MsoNormal"><b><span lang="EN-US" style="FONT-SIZE: 10pt">From:</span></b><span lang="EN-US" style="FONT-SIZE: 10pt"> <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>
<div class="im"><br><b>Subject:</b> Re: [cisco-voip] When is Transcoding Required?</div></span>
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<p class="MsoNormal">Hi Aaron,</p></div>
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<p class="MsoNormal"> <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.</p>
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<p class="MsoNormal">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.</p>
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<p class="MsoNormal">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.</p>
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<p class="MsoNormal">Use this link below about number of calls per DSP etc... High Complexity codec would mean G729 to name one.</p></div>
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<p class="MsoNormal"><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></p>
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<p class="MsoNormal">hope that helps.</p></div>
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<p class="MsoNormal">cheers,</p></div>
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<p class="MsoNormal">Daniel</p></div>
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<p class="MsoNormal"><br> </p></div>
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<p class="MsoNormal">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:</p>
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<p class="MsoNormal"><span style="FONT-SIZE: 9pt; COLOR: navy">Hi Guys,</span></p>
<p class="MsoNormal"><span style="FONT-SIZE: 9pt; COLOR: navy"> </span></p>
<p class="MsoNormal"><span style="FONT-SIZE: 9pt; 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></p>
<p class="MsoNormal"><span style="FONT-SIZE: 9pt; COLOR: navy"> </span></p>
<p class="MsoNormal"><span style="FONT-SIZE: 9pt; COLOR: navy">Thanks,</span></p>
<p class="MsoNormal"><span style="FONT-SIZE: 9pt; COLOR: navy"> </span></p>
<p class="MsoNormal"><span style="FONT-SIZE: 9pt; COLOR: navy">Aaron.</span></p></div>
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