[OpenSIPS-Users] reinviting to a recording server

[Tito Cumpen]

Eric,


I am going to be using the dialogic XMS. I believe it handles SAVPF(DTLS).
The request to record will be triggered by an api. Below is a diagram of
what I intend to do.


Bogdan,

I'd like to know if I can trigger a reinvite via the MI_http interface
which will enact a b2bua scenario with the intention to move both legs to
the media server. Below is a diagram. One detail I'd like to point out is
the blank reinvite needs to source from opensips as it carries all the
headers. Please advise if this is possible or if I can do anything aside
from using a b2bua scenario.






[image: Inline image 1]




On Wed, Sep 9, 2015 at 6:09 AM, Bogdan-Andrei Iancu <bogdan at opensips.org>
wrote:

> 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 Developerhttp://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 <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 Developerhttp://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
>>
>>
>> _______________________________________________
>> Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>>
>>
>
>
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