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
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