[cisco-voip] SIP Load and Re-Invite

[Dennis Heim]

Do you have a cube in place?

Dennis Heim
Network Voice Engineer
CDW  Advanced Technology Services
11711 N. Meridian Street, Suite 225
Carmel, IN  46032

317.569.4255 Single Number Reach
317.569.4201 Fax
dennis.heim at cdw.com<mailto:dennis.heim at cdw.com>
cdw.com/content/solutions/unified-communications/<http://www.cdw.com/content/solutions/unified-communications/>

From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Jason Aarons (AM)
Sent: Friday, April 15, 2011 12:51 PM
To: cisco-voip (cisco-voip at puck.nether.net)
Subject: [cisco-voip] SIP Load and Re-Invite

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?

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.

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