<html>
<head>
<meta content="text/html; charset=windows-1252"
http-equiv="Content-Type">
</head>
<body bgcolor="#FFFFFF" text="#000000">
<p>You can setup a transcoder at Site A and set the regions as
follows:</p>
<p>Site A <----> Site B (7kbps / G723)</p>
<p>Site A<---->Transcoder (G729 or higher)</p>
<p>Site B<---->Transcoder (G729 or higher)</p>
<p>Because the region config is set to G723, a transcoder will be
invoked. The Region config between the transcoders and each site
will be checked. If neither side is talking G711 to the
transcoder you will need a universal transcoder
(<a class="moz-txt-link-freetext" href="http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/rel_notes/7_1_5/cucm-rel_notes-715.html#wp1768778">http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/rel_notes/7_1_5/cucm-rel_notes-715.html#wp1768778</a>).
With CUCM 9.x+ you can accomplish something similar with Audio
Codec Preference Lists.<br>
</p>
<p>--<br>
Adam<br>
</p>
<p><br>
</p>
<div class="moz-cite-prefix">On 8/24/2016 3:29 AM, Mauro Celli
wrote:<br>
</div>
<blockquote
cite="mid:7E5F77F37136A9448D89426CCACB3F76C4D57791@HME-MBX01.hme.2000net.it"
type="cite">
<pre wrap="">Hi,
i have three site,
Cucm is in SiteA
SiteB and SiteC have a vpn to SiteA
SiteB and SiteC they can not have a vpn between them for safety reasons.
To make calls from SiteB and SiteC, i have to enable "use trusted relay
point" for all phone of B-C sites, and insert MTP.
But if i enable this TRP, when a SiteB phone call another SiteB phone, MTP
is inserted in call and i consume wan bandwith.
How i can limit trusted relay point only when it is really necessary? I have
CUCM 8.6
Thanks
</pre>
<br>
<fieldset class="mimeAttachmentHeader"></fieldset>
<br>
<pre wrap="">_______________________________________________
cisco-voip mailing list
<a class="moz-txt-link-abbreviated" href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a>
<a class="moz-txt-link-freetext" href="https://puck.nether.net/mailman/listinfo/cisco-voip">https://puck.nether.net/mailman/listinfo/cisco-voip</a>
</pre>
</blockquote>
<br>
</body>
</html>