[cisco-voip] SIP Load and Re-Invite

[Mike Lydick]

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> 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 Single Number Reach
> 317.569.4201 Fax
> 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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