We'd like to use a Freeswitch server as SIP communication server connecting to our mobile client apps directly. Regarding to your web site Freeswitch is a supported server component to your VOIP SDK.
In our test environment we faced the following problem. After the successful connection between the two mobile clients and the call initialized the call is disconnected after a certain timeout. After examining the Freeswitch output and the Freeswitch documentation the cause was revealed: Aduio connection initialization failure. Freeswitch needs a large open port range (16384-32768 UDP). With the server located in a protected NAT environment forwarding this range to one server is not a suitable solution.
Could you provide us a safe and working configuration example of that setup? Currently the only goal is to let two clients from the public internet to connect and speak with each other.
(We also see that Freeswitch has NAT traversal options using a STUN server, however configuring a STUN server is not straightforward to us at the moment. If this is the only way: Could you help us with information on STUN configuration?)
Thanks in advance!