[SIPForum-discussion] SIP UAs that allow in-call SDP param changes?
Anthony Orlando
avorlando at yahoo.com
Sat Jan 5 09:13:12 UTC 2008
You say you'd like to change codec preferences. Would
a codec change itself work for you? If so do you have
a couple of ATA's and faxes? You could set the ATA's
up for a low bit rate codec with g.711 fallback. Make
a call with the fax handset then send a fax with the
call still up. You would obviously need a fax on the
other end as well.
What should happen is that the call is established
with the low bit rate codec then when the ATA detects
a fax it should change to G.711.
Anthony
--- Jeff Wright <JWright at azteknetworks.net> wrote:
> Does anyone know of a SIP UA that allows for in-call
> SDP parameter
> changes? For instance, I'd like to be able to
> change codecs (via a
> re-INVITE and subsequent renegotiation) while a call
> is up.
>
>
>
> My overall motivation is to be able to run a test to
> force a call flow
> something akin to Example 3.7 in RC3665
> <http://tools.ietf.org/html/rfc3665> . That example
> shows an IP address
> change, but I was hoping that I could similarly
> force a re-INVITE to
> occur if I changed codec preferences in a UA while a
> call was up. So
> far, I've not found a UA that supports this (ZoIPer
> and X-Lite both
> allow me to change the preferred codecs while a call
> is up but they
> don't renegotiate with my new choice). I also tried
> changing the IP
> address of my PC in the middle of the call, but
> neither call agent does
> anything about that, either.
>
>
>
> Jeffrey D. Wright
>
> Aztek Networks, Inc.
>
>
>
>
>
>
>
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