I’ve been looking at some off the shelf gateways and am exploring the possibility of some cost saving by implementing an open source solution.
What we have is a private SIP network operating with a software VoIP system. And what I’ve specifically been looking at is a way of integrating another webRTC based software package into this network. So just a conversion between webRTC and SIP (assuming for the moment that the codecs used are the same each side).
I think as a proof of concept it should be able to handle about 150 simultaneous links (with a view to scale up).
Would anybody be able to advise me firstly whether Asterisk is appropriate for this purpose? And what sort of hardware would be required to run it? I’m quite new to this technology so any advice would be appreciated.