#1 2012-06-05 11:20:24

**_kevinmui_**

sip2sip for FMG

Has anyone successfully used sip2sip for universal voice?  I'm pretty confused about the whole Gateway thing!

**_mrock66_**

Re: sip2sip for FMG

You figure this out yet?

**_kevinmui_**

Re: sip2sip for FMG

Nope, never got this to work :(

**_praha_**

Re: sip2sip for FMG

@kevinmui: Universal Voice (UV) is required only if you are using one of these features:

-Broadcast telephony audio conference to participants
-InterCall telephony adaptor for recording (audio)
-Arkadin telephony adaptor for recording (audio)
-Using a non-integrated telephony audio conference solution where you want to record audio

In order to use the features mentioned above, you will need to have Flash Media Gateway (FMG) and SIP.

**_irfant_**

Re: sip2sip for FMG

We, at Clemson University, are in late stages of testing rPhone/ZipDX (http://www.refineddata.com) based Telephony solutions for our Adobe Connect deployment, potentially replacing  the existing PGi-based solution. rPhone/ZipDX offer their own SIP Server and that has made possible audio from all sources to be part of Connect teleconferencing sessions, including saving into Recordings. After some more tests we will be ready to deploy the new telephony solution.
Also, if most users use their computer's audio (microphones) and only a few users dial-in then rPhone/ZipDX should dramatically decrease the $ amount for teleconferencing sessions!
Finally, being a Connect Sys Admin, rPhone/ZipDX makes management of the telephony configuration much easier than PGi: No need to have any telephony .conf files etc configured (which often get over-written when the server is patched/updated/upgraded), probably no need to even run the Telephony Service; FMG is already a part of the Connect Pro deployment so no worries about that either.
Irfan