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<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D">On the PVDM2 models of DSPs, conferencing reserves an entire DSP. I thought that the new PVDM3 modules with the new ISR G2 routers overcome this limitation.<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D"><o:p> </o:p></span></p>
<p class="MsoNormal"><b><span style="font-family:"Arial","sans-serif";color:black">Q.</span></b><span style="color:black"> Can conferencing share the resources with transcoding or voice calls of a single DSP?<o:p></o:p></span></p>
<p class="MsoNormal"><a name="wp9000103"></a><b><span style="font-family:"Arial","sans-serif";
color:black">A.</span></b><span style="color:black"> No. Conferencing needs a dedicated DSP resource. If a DSP is assigned for a conferencing session, then it cannot
be used for transcoding or voice call initiations or terminations. Transcoding and voice calls can share the resource of a single DSP, however. Note that conferencing needs a dedicated DSP, but not a dedicated PVDM2 module. For example, PVDM2-64 contains 4
DSPs; if one of them is used for conferencing, the other three can still be used for other purposes.<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D"><o:p> </o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D">http://www.cisco.com/en/US/prod/collateral/modules/ps3115/ps6024/prod_qas0900aecd8016c6ad_ps5855_Products_Q_and_A_Item.html<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D"><o:p> </o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D">Bret<o:p></o:p></span></p>
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<p class="MsoNormal"><b><span style="font-size:10.0pt;font-family:"Tahoma","sans-serif"">From:</span></b><span style="font-size:10.0pt;font-family:"Tahoma","sans-serif""> cisco-voip-bounces@puck.nether.net [mailto:cisco-voip-bounces@puck.nether.net]
<b>On Behalf Of </b>Syed Khalid Ali<br>
<b>Sent:</b> Wednesday, May 12, 2010 2:52 PM<br>
<b>To:</b> Ed Leatherman<br>
<b>Cc:</b> cisco-voip@puck.nether.net<br>
<b>Subject:</b> Re: [cisco-voip] HW Based Conferencing Bridge with 2xPVDM2-64<o:p></o:p></span></p>
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<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal">yes I tried the DSP calculator and for 16 conferencing session, a PVDM64 and PVDM32 is required.<o:p></o:p></p>
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<p class="MsoNormal"><o:p> </o:p></p>
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<p class="MsoNormal" style="margin-bottom:12.0pt">But I am confused, the same DSPs can be used for both Voice Termination and Conferencing.<o:p></o:p></p>
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<p class="MsoNormal">On Thu, May 13, 2010 at 12:11 AM, Ed Leatherman <<a href="mailto:ealeatherman@gmail.com">ealeatherman@gmail.com</a>> wrote:<o:p></o:p></p>
<p class="MsoNormal">Syed,<br>
<br>
Have you tried the online DSP calculator at:<br>
<a href="http://www.cisco.com/cgi-bin/Support/DSP/dsp-calc.pl" target="_blank">http://www.cisco.com/cgi-bin/Support/DSP/dsp-calc.pl</a><o:p></o:p></p>
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<p class="MsoNormal"><br>
<br>
<br>
On Wed, May 12, 2010 at 3:05 PM, Syed Khalid Ali<br>
<<a href="mailto:syed.khalid.khursheed@gmail.com">syed.khalid.khursheed@gmail.com</a>> wrote:<br>
> Group:<br>
> I need some suggestion.<br>
> Currently we are running CCM 4.2. We have around 30+ site with 5 to 10<br>
> phones. To save bandwidth and limitation, regions and location have been<br>
> implemented. G.729 is used for inter-site calls and G.711 for intra-site<br>
> calls. Currently the voice gateway is placed at head office with two E1s and<br>
> 12 FXOs. The issue is conferencing with inter-site calls due to G.729.<br>
> There are 2 PVDM64 installed. 60 are consumed by E1s and 12 by FXOs total of<br>
> 72. I still have (128-72) 76 channels. Is the rest of channels can be used<br>
> for hardware based conferencing. And how many session will be available.<br>
> Can any one provide a good resource for doing this math?<br>
> Also since we don't have voice mail. I guess there is no need for<br>
> transcoding.<br>
> --<br>
> Thanks,<br>
> Syed Khalid Ali<br>
><br>
><o:p></o:p></p>
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<p class="MsoNormal">> _______________________________________________<br>
> cisco-voip mailing list<br>
> <a href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a><br>
> <a href="https://puck.nether.net/mailman/listinfo/cisco-voip" target="_blank">https://puck.nether.net/mailman/listinfo/cisco-voip</a><br>
><br>
><br>
<br>
<br>
<br>
--<br>
<span style="color:#888888">Ed Leatherman</span><o:p></o:p></p>
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<p class="MsoNormal" style="margin-bottom:12.0pt"><br>
<br clear="all">
<br>
-- <br>
Thanks,<br>
Syed Khalid Ali<o:p></o:p></p>
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