Send discussion mailing list submissions to
discussion sipforum.org
To subscribe or unsubscribe via the World Wide Web, visit
http:
//sipforum.org/mailman/listinfo/discussion
or, via email, send a message with subject or body 'help'
to
discussion-requestsipforum.org
You can reach the person managing the list at
discussion-ownersipforum.org
When replying, please edit your Subject line so it is more
specific
than "Re: Contents of discussion digest..."
Today's Topics:
1. Re: SIP UAs that allow in-call SDP param changes?
(Anthony Orlando)
2. (no subject) (Naresh Mehta)
3. Re: SIP UAs that allow in-call SDP param changes?
(Jeff Wright)
------------------------------------------------------------
----------
Message: 1
Date: Sat, 5 Jan 2008 01:13:12 -0800 (PST)
From: Anthony Orlando <avorlandoyahoo.com>
Subject: Re: [SIPForum-discussion] SIP UAs that allow
in-call SDP
param changes?
To: Jeff Wright <JWrightazteknetworks.net>,
discussionsipforum.org
Message-ID: <253666.13909.qmweb51004.mail.re2.yahoo.com>
Content-Type: text/plain; charset="iso-8859-1"
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 <JWrightazteknetworks.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.i
etf.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.
>
>
>
>
>
>
>
> > _______________________________________________
> This is the SIP Forum discussion mailing list
> TO UNSUBSCRIBE, or edit your delivery options,
> please visit
> http:
//sipforum.org/mailman/listinfo/discussion
> Post to the list at discussionsipforum.org
>
____________________________________________________________
________________________
Be a better friend, newshound, and
know-it-all with Yahoo! Mobile. Try it now. http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9
tAcJ
------------------------------
Message: 2
Date: Sat, 5 Jan 2008 07:30:44 -0800 (PST)
From: Naresh Mehta <naresh27_mehtayahoo.com>
Subject: [SIPForum-discussion] (no subject)
To: discussionsipforum.org
Message-ID: <37781.77965.qmweb44802.mail.sp1.yahoo.com>
Content-Type: text/plain; charset=iso-8859-1
Hello everybody,
I am working on project for
"Provisioning of SIP Server". For this I need
material
regarding installing,configuring and activation of SIP
based components.Suggest me to which all software
components I need to install since there many such
components available.Now each component has its own
configuration, can anyone guide me how move on.
____________________________________________________________
________________________
Be a better friend, newshound, and
know-it-all with Yahoo! Mobile. Try it now. http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9
tAcJ
------------------------------
Message: 3
Date: Sat, 5 Jan 2008 11:21:02 -0500
From: "Jeff Wright" <JWrightazteknetworks.net>
Subject: Re: [SIPForum-discussion] SIP UAs that allow
in-call SDP
param changes?
To: "Anthony Orlando" <avorlandoyahoo.com>, <discussionsipforum.org>
Message-ID:
<ABFF960C9FAE494482BD527AB5D6263F103C24MI8NYCMAIL03.Mi8.com>
Content-Type: text/plain; charset="iso-8859-1"
That's not a bad idea. I do have a couple of ATAs in our
test lab and I'll see which ones support fax (either T.38 or
G.711 passthrough). Thanks for the tip.
Jeffrey Wright
System Test Engineering Manager
Aztek Networks
-----Original Message-----
From: Anthony Orlando [mailto:avorlandoyahoo.com]
Sent: Sat 1/5/2008 2:13 AM
To: Jeff Wright; discussionsipforum.org
Subject: Re: [SIPForum-discussion] SIP UAs that allow
in-call SDP param changes?
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 <JWrightazteknetworks.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.i
etf.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.
>
>
>
>
>
>
>
> > _______________________________________________
> This is the SIP Forum discussion mailing list
> TO UNSUBSCRIBE, or edit your delivery options,
> please visit
> http:
//sipforum.org/mailman/listinfo/discussion
> Post to the list at discussionsipforum.org
>
____________________________________________________________
________________________
Be a better friend, newshound, and
know-it-all with Yahoo! Mobile. Try it now. http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9
tAcJ
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://sipforum.org/p
ipermail/discussion/attachments/20080105/75a03866/attachment
-0001.html
------------------------------
_______________________________________________
This is the SIP Forum discussion mailing list
TO UNSUBSCRIBE, or edit your delivery options, please visit
http:
//sipforum.org/mailman/listinfo/discussion
Post to the list at discussionsipforum.org
End of discussion Digest, Vol 30, Issue 18
******************************************
|
