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<div class=Section1>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>All kidding aside SIP is moving towards supporting renegotiation
better. I sit on most of the SIP IETF lists (if you want some people that can
go into weeds on details – watch those lists) and a great number of
conversations lately revolve around this very subject.<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'>While many systems today loosely support renegotiation you can
expect your mileage to vary until the standards reach a better level of detail
and specificity. As I’ve been coming to learn lately (especially when it
comes to VoIP and Video) – SIP related RFC’s can have quite a wide
array of meanings even when the creator thinks they are being specific. I’ve
seen whole conversations on the IETF lists that pertain to proverbially dotting
I’s and crossing T’s because of how inexact some of the RFC’s
are.<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'>Expect the situation to improve – but in the meantime pick
and test your vendors carefully.<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>
<div>
<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;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>Jason Aarons
(AM)<br>
<b>Sent:</b> Friday, April 15, 2011 7:44 PM<br>
<b>To:</b> Mike Lydick; Dennis Heim<br>
<b>Cc:</b> cisco-voip (cisco-voip@puck.nether.net)<br>
<b>Subject:</b> Re: [cisco-voip] SIP Load and Re-Invite<o:p></o:p></span></p>
</div>
</div>
<p class=MsoNormal><o:p> </o:p></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>I think H323 v7 is due out soon to replace SIP </span><span
style='font-size:11.0pt;font-family:Wingdings;color:#1F497D'>J</span><span
style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>
Replaces ASN.1 with Creole.<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'>Mwen palé on ti kal Kreyol<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-size:10.0pt;font-family:"Tahoma","sans-serif"'>From:</span></b><span
style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'> Mike Lydick
[mailto:mike.lydick@gmail.com] <br>
<b>Sent:</b> Friday, April 15, 2011 2:19 PM<br>
<b>To:</b> Dennis Heim<br>
<b>Cc:</b> Jason Aarons (AM); cisco-voip (cisco-voip@puck.nether.net)<br>
<b>Subject:</b> Re: [cisco-voip] SIP Load and Re-Invite<o:p></o:p></span></p>
<p class=MsoNormal><o:p> </o:p></p>
<p class=MsoNormal>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".<o:p></o:p></p>
<div>
<p class=MsoNormal><o:p> </o:p></p>
</div>
<div>
<p class=MsoNormal>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.<o:p></o:p></p>
<div>
<p class=MsoNormal><o:p> </o:p></p>
</div>
<div>
<p class=MsoNormal>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.<o:p></o:p></p>
</div>
<div>
<p class=MsoNormal><o:p> </o:p></p>
</div>
<div>
<div>
<p class=MsoNormal>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.<o:p></o:p></p>
</div>
<div>
<p class=MsoNormal><o:p> </o:p></p>
</div>
<div>
<p class=MsoNormal>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?<o:p></o:p></p>
<div>
<p class=MsoNormal><o:p> </o:p></p>
</div>
<div>
<p class=MsoNormal>This SIP stuff is a fad anyways I am going to tell the
customer to move back to H323...<o:p></o:p></p>
<div>
<p class=MsoNormal><o:p> </o:p></p>
</div>
<div>
<p class=MsoNormal style='margin-bottom:12.0pt'><br clear=all>
Best Regards,<br>
<br>
Mike Lydick<br>
<br>
<br>
<o:p></o:p></p>
<div>
<p class=MsoNormal>On Fri, Apr 15, 2011 at 1:35 PM, Dennis Heim <<a
href="mailto:Dennis.Heim@cdw.com">Dennis.Heim@cdw.com</a>> wrote:<o:p></o:p></p>
<div>
<div>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span
style='color:#1F497D'>Do you have a cube in place?</span><o:p></o:p></p>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span
style='color:#1F497D'> </span><o:p></o:p></p>
<div>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><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" target="_blank">317.569.4255</a> Single Number Reach<br>
<a href="tel:317.569.4201" 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><o:p></o:p></p>
</div>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span
style='color:#1F497D'> </span><o:p></o:p></p>
<div>
<div style='border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0in 0in 0in'>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><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><o:p></o:p></p>
</div>
</div>
<div>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'> <o:p></o:p></p>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>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?<o:p></o:p></p>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'> <o:p></o:p></p>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>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.<o:p></o:p></p>
<p class=MsoNormal style='mso-margin-top-alt:auto;margin-bottom:12.0pt'> <o:p></o:p></p>
<div class=MsoNormal align=center style='text-align:center'>
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