[SIPForum-discussion] Codec Changes during Call

[M. Ranganathan]

On Wed, Dec 24, 2008 at 8:51 AM, Manish Jain <jain.manish30 at gmail.com> wrote:
> Merry Christmas to All,
>
> I want to know, is there a way to change the codecs during the call between
> two SIP Phones.

Yes there is. You can do an in-dialog re-INVITE with a new SDP
specifying new codec preferences.

>
> If yes , then
>
> What is the use/Requirements of changing the codecs and with which
> application it is required?


Typically, call transfer from one phone to another. Another use case
is fax over IP. For Fax, the call setup will typically begin with G711
and switch to t-38 after handshake.

> Codecs changes is done by network ?
> Is it acceptable at all in the when designing the network?

Huh? The network is a passive entity. Endpoints initiate such things.

>
> --
> With Regards,
> Manish Jain
>
> Luck Is Like a Blank Paper,where we writes our Future.
>
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-- 
M. Ranganathan