[Kamailio-Users] re-invite mid call?

Antonio Goméz Soto antonio.gomez.soto at gmail.com
Fri Jan 1 17:32:44 CET 2010


Op 31-12-09 09:42, Alex Balashov schreef:
> No.  It's a proxy, it can't originate requests.
>

I am sorry, maybe I do not understand enough about SIP and proxies.
Wouldn't it be possible to fake re-invites? Tell each phone that it
gets a reinvite coming from the other phone, and basically redirecting
the sound stream to an rtpproxy for example? or a codec translator?

I'd like to switch from g711 to g729 when the link gets overloaded
for example. Or record the phonecall.

I am trying to learn more about SIP. Aren't reinvites within
a dialog common?

Maybe I could make changes to openser to enable this?

Antonio



> On 12/31/2009 03:33 AM, Antonio Goméz Soto wrote:
>
>> Hi,
>>
>> maybe I should go to the -dev list for this, but thought I'd ask here
>> first.
>> Is it possible to let openSER send a SIP reinvite in the middle of a
>> call,
>> to redirect the sound stream to some rtpproxy? Maybe using the MI?
>>
>> Or should OpenSER be modified to do this?
>>
>> Thanks,
>> Antonio
>>
>>
>> _______________________________________________
>> Kamailio (OpenSER) - Users mailing list
>> Users at lists.kamailio.org
>> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users
>> http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
>
>





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