[cisco-voip] SIP Load and Re-Invite

[Jason Aarons (AM)]

I think H323 v7 is due out soon to replace SIP :) Replaces ASN.1 with Creole.

Mwen palé on ti kal Kreyol

From: Mike Lydick [mailto:mike.lydick at gmail.com]
Sent: Friday, April 15, 2011 2:19 PM
To: Dennis Heim
Cc: Jason Aarons (AM); cisco-voip (cisco-voip at puck.nether.net)
Subject: Re: [cisco-voip] SIP Load and Re-Invite

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".

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.

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.

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.

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?

This SIP stuff is a fad anyways I am going to tell the customer to move back to H323...


Best Regards,

Mike Lydick



On Fri, Apr 15, 2011 at 1:35 PM, Dennis Heim <Dennis.Heim at cdw.com<mailto:Dennis.Heim at cdw.com>> wrote:
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<tel:317.569.4255> Single Number Reach
317.569.4201<tel: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> [mailto: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<mailto: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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