Hi Tito,
A SIP call can have only 2 end-points. What is not clear for me is:
after inserting the media server, what is the final configuration in
terms of who's talking to who? Still A talks to B, but media server is
recording ? or A talks to media server (like VM) and B drops out ?
Best regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 08.09.2015 21:27, Tito Cumpen wrote:
Bogdan,
Thanks for your reply and questions. Currently call flows are using
ICE and rtpengine as a turn relay and so there's nothing in between .
In the case I get a request to begin recording I'd like to move the
active call to a media server that bridges the call making it appear
seamless for the caller and callee. If I trigger a RE-INVITE to both A
and B with the media server address this should work but I am not sure
how I can use opensips to send a blank invite on behalf of both A and
B utilizing the same call id to media server then utilizing the reply
as the RE-INVITE to A and B. In essence putting the media server in
between without forcing a hang up.
Thanks,
Tito
On Mon, Sep 7, 2015 at 6:20 AM, Bogdan-Andrei Iancu
<bog...@opensips.org <mailto:bog...@opensips.org>> wrote:
Hi Tito,
Do you want to move on the call legs to the call recording server
(like to a VM or so) or while A talks to B, you want to have
something in the middle to record the call between those two parties ?
Best regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 03.09.2015 01:13, Tito Cumpen wrote:
Group,
Has anyone had experience reinviting an ongoing session between
two sip clients to a sip capable media server for call recording
purposes without dropping the ongoing call? Is the best practice
to use XML_RPCNG/fifo command and have opensips interact as 3rd
party call control. Or would the 3rd party entity need to hijack
the ongoing session as pose as the remote party. I have a
requirement to record video and audio legs. The media server is
capable for recording these streams just need to find a way to do
this without dropping the call.
Thanks,
Tito
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