[cisco-voip] SIP Load and Re-Invite

[Mike Lydick]

Thanks Ryan,

I believe we have design incompatibility based on your comment. We are using
media flow-around so I guess that would remove the DSP out of the RTP stream
so to speak unless we can dynamically allocate one after call setup. I agree
the IP phone would not be the correct test so we are moving to a FXS port on
our testing. I will removed the media flow-around to see if this helps but
flow-around is a requirement as we will have 500 sites and do not intend of
pinning all the RTP traffic to the Cube.

I am fussier on the comment about changing codec in a way we can act on it,
do you mean a endpoint that can do something with the fax tone.

Oh sorry for the typo, CNG not CMG...

Best Regards,

Mike Lydick




On Fri, Apr 15, 2011 at 2:49 PM, Ryan Ratliff <rratliff at cisco.com> wrote:

> re what the point is in a 79XX phone responding to something like a fax
> tone since it couldn't do anything with said fax data anyway.
>
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