[OpenSIPS-Users] reinviting to a recording server

[Eric Tamme]

Hi Tito,

I am curious about your situation because I am in the early stages of 
doing call recording as well.  I have some questions if you do not mind.

Are you doing DTLS-SRTP and or SDES-SRTP, or are you simply using ICE 
and rtpengine is a turn relay?  If you are not doing DTLS or ICE, 
perhaps you should look into rtpproxy which does have a recording 
feature.  If you are doing DTLS, is your recording server DTLS capable - 
and if so would you mind telling me what you are using for a recording 
server?

I am also curious how you are detecting a recording request - is it via 
DTMF, or some external signalling mechanism like clicking record on a 
web interface?  If you are doing DTMF detection, I'd like to know how 
you are doing it.

-Eric

What are you using as the recording server - also are you doing DTLS 
with rtpengine or just ICE, I would also like to know how you plan to 
trigger the detection of a recording request - are you

On 09/08/2015 12:27 PM, 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 
> <bogdan at opensips.org <mailto:bogdan at 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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>
>
>
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