I am new to voip and asterisk and I not sure if my statement below is required or not.
If I create a few extensions for my friends on my asteriskNow server , do I need to open a port ( Port Forwarding ) on my router so they can register their softphones on my asteriskNow server?
If the above is not required, then how do my friends register their softphones to my asteriskNow server to make free calls?
For SIP there are two types of traffic that need to be forwarded: SIP signaling and RTP media.
The default port for udp based SIP signaling is port 5060. Nevertheless, you will still need to check your PBX to find out what port it is using.
The RTP media traffic (the actual audio stream) uses a range of udp ports that varies greatly from PBX to PBX and is usually configurable. A typical range might be 10000-20000. However, you will only need to utilize a range that is large enough to support the number of simultaneous udp ports you plan to have. So in the case of port forwarding, it makes sense to configure your PBX with as small a range as makes sense - say 10000-10100 - and forward only to those ports