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</o:shapelayout></xml><![endif]--></head><body lang=EN-US link=blue vlink=purple><div class=WordSection1><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><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'><hr size=1 width="100%" align=center></div><p><strong><span style='font-size:10.0pt;color:gray'>Disclaimer: This e-mail communication and any attachments may contain confidential and privileged information and is for use by the designated addressee(s) named above only. If you are not the intended addressee, you are hereby notified that you have received this communication in error and that any use or reproduction of this email or its contents is strictly prohibited and may be unlawful. If you have received this communication in error, please notify us immediately by replying to this message and deleting it from your computer. Thank you. </span></strong><o:p></o:p></p></div></div></div><p class=MsoNormal style='margin-bottom:12.0pt'><br>_______________________________________________<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><o:p></o:p></p></div><p class=MsoNormal><o:p> </o:p></p></div></div></div></div></div></div></body></html>