I'm glad you asked this as a G711 call its much simpler<br> <br>I would say one Channel, this is how I understand it<br> <br>1 DSP has 16 channels which allows you to have 1 G711 call per channel for packetisation so 16 G711 calls.<br>
<br> <br> Use this command "sh voice dsp group all"<br> <br>output<br> <br>DSP groups on slot 0:<br>dsp 1:<br> State: UP, firmware: 4.4.25<br> Max signal/voice channel: 16/16<br> Max credits: 240<br> Group: FLEX_GROUP_VOICE, complexity: FLEX<br>
Shared credits: 180, reserved credits: 0<br> Signaling channels allocated: 16<br> Voice channels allocated: 3<br> Credits used: 60<br> Voice channels:<br> Ch01: voice port: 0/1/0:15.31, codec: g711ulaw, credits allocated: 15<br>
Ch02: voice port: 0/1/0:15.1, codec: g729r8, credits allocated: 30<br> Ch03: voice port: 0/1/0:15.30, codec: g711ulaw, credits allocated: 15<br> <br>This output is looking at 3 calls from the ISDN two G711 calls to site phones one G729 call to unity connection, this is packetisation from ISDN to IP not transcoding. From this output you can see that a G711 call takes 15 credits which is the same as 1 channel. 15 credits times 16 channels is 240 available credits. Credits for a G729 call are 30 so double the needed amount of Channels which means more DSP's. Your shared credits is whats left over 15 + 15 + 30 - 240 = 180. <br>
<br>hope this helps<br><br>
<div class="gmail_quote">On Fri, Dec 11, 2009 at 1:46 PM, Lelio Fulgenzi <span dir="ltr"><<a href="mailto:lelio@uoguelph.ca">lelio@uoguelph.ca</a>></span> wrote:<br>
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<div style="FONT-SIZE: 10pt; COLOR: #000000; FONT-FAMILY: Verdana">ok, here's a question for you.....if i'm taking a call from the pstn to an ip phone using g711, am i using one channel on the dsp or two?<br><br>
<br><br>---<br>Lelio Fulgenzi, B.A.<br>Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1<br>(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)<br>^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^<br>
"Bad grammar makes me [sic]" - Tshirt
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<div class="h5"><br><br><br>----- Original Message -----<br>From: "Daniel" <<a href="mailto:dan.voip@danofive.id.au" target="_blank">dan.voip@danofive.id.au</a>><br>To: "Aaron Riemer" <<a href="mailto:ariemer@wesenergy.com.au" target="_blank">ariemer@wesenergy.com.au</a>><br>
Cc: <a href="mailto:cisco-voip@puck.nether.net" target="_blank">cisco-voip@puck.nether.net</a><br>Sent: Thursday, December 10, 2009 10:32:22 PM GMT -05:00 US/Canada Eastern<br>Subject: Re: [cisco-voip] When is Transcoding Required?<br>
<br>
<div>Hi Aaron,</div>
<div> <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.</div>
<div> </div>
<div>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.</div>
<div> </div>
<div>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.</div>
<div> </div>
<div>Use this link below about number of calls per DSP etc... High Complexity codec would mean G729 to name one.</div>
<div><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></div>
<div> </div>
<div>hope that helps.</div>
<div> </div>
<div>cheers,</div>
<div> </div>
<div>Daniel</div>
<div> </div>
<div><br> </div>
<div class="gmail_quote">On Fri, Dec 11, 2009 at 12:36 PM, Aaron Riemer <span dir="ltr"><<a href="mailto:ariemer@wesenergy.com.au" target="_blank">ariemer@wesenergy.com.au</a>></span> wrote:<br>
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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><br>LEGAL DISCLAIMER: This message contains confidential information and is intended only for the individual named. If you are not the named addressee you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately by e-mail if you have received this e-mail by mistake and delete this e-mail from your system. If you are not the intended recipient you are notified that disclosing, copying, distributing or taking any action in reliance on the contents of this information is strictly prohibited.<br>
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