As I understand, the communication between SIP phone and Asterisk server is as follows:
- Upon initial communication, the SIP phone sends a media port range to Asterisk.
- Asterisk picks two random ports out of this range and initiates audio conversion on these ports.
I noticed that Cisco SIP phones have a default media port range of 16384 to 32766. Why can’t I just narrow it down to two ports? For example:
SIP phone 1 - 16384-16385
SIP phone 2 - 16386-16387
and so on…
In our case, the phones are behind NAT. With the above setup, all I would need to do is define a simple port range forwarding on the router.
UDP port 16384-16385 ==> Forward to the IP of SIP phone 1
UDP port 16386-16387 ==> Forward to the IP of SIP phone 2
Does anyone see a problem with this?
Thank you in advance for your help.