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cwienands Intermediate
Joined: April 02 2013 Location: United States Posts: 4
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Posted: April 23 2013 at 11:33am | IP Logged
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Hello,
we are currently trying to understand better whether the MediaEngine will fulfill our needs for a conferencing system.
In this system, a SIP Server such as FreeSwitch or SIP-capable "hardware" appliance will run. This SIP Server is sized to handle calls and conferencing for hundreds of connections. Our vision was to use a SIP SDK and a "remote control app" to essentially remote control dial-outs, adding and removing calls from conference sessions, muting and unmuting individual calls within conference sessions, etc. Judging by the SIP RFCs, this should be possible. Since our system would just be remote-controlling calls but not processing/generating audio streams, the load caused by the "remote control app" would be very small.
I studied the SW Developer's Reference for the LanScape VoIP Media Engine and I am not sure if it is doing what we want. I am under the impression that all the calls and audio streams from/to external phones are forwarded to the Media Engine, and that the Media Engine does all the conferencing, mixing of audio streams, muting/unmuting, etc. internally.
Coming back to our vision... if my assumption about the MEdia Engine are correct then the highly performant SIP Server would only be used to pass calls through, and therefore the machine running the Media Engine would have be properly sized as well to handle the call volume, and there actually might be additional load on the main SIP Server because every call to/from the outside is forwarded to the Media Engine.
Can you please clarify how the Media Engine works, and whether it is possible to make use of the conferencing capabilities in a connected SIP Server?
Thank you very much, Christoph
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support Administrator
Joined: January 26 2005 Location: United States Posts: 1666
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Posted: April 24 2013 at 10:29am | IP Logged
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Christoph,
Your understanding of our VOIP SDK seems accurate. Our VOIP SDK can be used to develop multiple different types of VOIP applications depending on your specific needs and what VOIP behaviors we have implemented in the product.
If you want to develop your own conferencing server, you can use our SDK to do that. You can also determine how your “remote control app” will then control your conferencing server app that is based on our VOIP SDK. In this case, if you have a “front end” VOIP device/PBX, it will simply be used to pass calls to your conferencing server as needed.
If you want to use our VOIP SDK to attempt to control conference sessions on some other external piece of VOIP equipment, it may be possible. I can’t say for certain how you would do this seeing I do not know any specifics regarding available communication protocols or APIs offered by the other VOIP device (FreeSwitch, "hardware" appliance, etc).
Please see the “Software Developer’s Reference” for complete information pertaining to how the SDK functions. Its all there…
If you want or need assistance designing your eventual system, we do offer paid support simply for the asking.
RJ
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