The field test was Verizon introducing CMG tones a few seconds after the call was established to a Cisco 7931 phone. The Cube logged: "Failed to negotiate media: 488".<div><br></div><div>In our example we have CUBE and CM8.5 in place but the call fails to switch the Media stream codec after a CMG tone is sent.<div>
<br></div><div>We are not using 'Require MTP', but tried the Dynamic MTP option in the SIP Profile on UCM but the call still fails after the codec change is sent and this option appears to alway enable the MTP.</div>
<div><br></div><div><div>Tac is questioning if this supported and what scenario would require this. The only one that I can think is an Analog port that has a multifunction Fax, or a line that is being shared for voice an modem. We are going to retest on a FXS port on Monday.</div>
<div><br></div><div>I see an option for modifying SDP for mid call codec changes, we did not enable at the time of testing. Do not have any documentation that states that this is required?<div><br></div><div>This SIP stuff is a fad anyways I am going to tell the customer to move back to H323...<br>
<div><br></div><div><br clear="all">Best Regards,<br><br>Mike Lydick<br><br><br>
<br><br><div class="gmail_quote">On Fri, Apr 15, 2011 at 1:35 PM, Dennis Heim <span dir="ltr"><<a href="mailto:Dennis.Heim@cdw.com">Dennis.Heim@cdw.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">
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<p class="MsoNormal"><span style="color:#1F497D">Do you have a cube in place?</span></p>
<p class="MsoNormal"><span style="color:#1F497D"> </span></p>
<div>
<p class="MsoNormal"><span style="font-size:10.0pt;color:#1F497D">Dennis Heim<br>
Network Voice Engineer<br>
CDW Advanced Technology Services<br>
11711 N. Meridian Street, Suite 225<br>
Carmel, IN 46032<br>
<br>
<a href="tel:317.569.4255" value="+13175694255" target="_blank">317.569.4255</a> Single Number Reach</span><span style="font-size:10.0pt;color:#1F497D"><br>
<a href="tel:317.569.4201" value="+13175694201" target="_blank">317.569.4201</a> Fax</span><span style="color:#1F497D"> <br>
<a href="mailto:dennis.heim@cdw.com" target="_blank"><span style="font-size:10.0pt">dennis.heim@cdw.com</span></a></span><u><span style="color:blue"><br>
</span></u><span style="color:#1F497D"><a href="http://www.cdw.com/content/solutions/unified-communications/" target="_blank">cdw.com/content/solutions/unified-communications/</a></span></p>
</div>
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<div style="border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0in 0in 0in">
<p class="MsoNormal"><b><span style="font-size:10.0pt">From:</span></b><span style="font-size:10.0pt"> <a href="mailto:cisco-voip-bounces@puck.nether.net" target="_blank">cisco-voip-bounces@puck.nether.net</a> [mailto:<a href="mailto:cisco-voip-bounces@puck.nether.net" target="_blank">cisco-voip-bounces@puck.nether.net</a>]
<b>On Behalf Of </b>Jason Aarons (AM)<br>
<b>Sent:</b> Friday, April 15, 2011 12:51 PM<br>
<b>To:</b> cisco-voip (<a href="mailto:cisco-voip@puck.nether.net" target="_blank">cisco-voip@puck.nether.net</a>)<br>
<b>Subject:</b> [cisco-voip] SIP Load and Re-Invite</span></p>
</div>
</div><div class="im">
<p class="MsoNormal"> </p>
<p class="MsoNormal">Is it normal for sip providers (say Verizon) to want to change codec mid-call or require your equipment can do it? I understand CallManager 8.5 / SIP 7945 can’t do this. Is it a CallManager limitation or a phone load limitation or both
for reinvite to change codec mid-call?</p>
<p class="MsoNormal"> </p>
<p class="MsoNormal">I’m a fan of codecs that dynamically change bandwidth (Silk, iLBC) but not G.722 to G711 to G.729 if the call degrades, but I guess with SIP you could possible do this based on the spec for re-invite. Or switch from G.729 to G.711 for faxes,
etc.</p>
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