Although it is troublesome to use stun server for Asterisk 1.6.2.11, voip-info.org does recommend to drop this feature in the sip.conf.
I followed the recommendation and found that one way audio problem encountered, i.e., the outbound audio is going through the firewall but not the inbound one. Put the stun server in the sip.conf, the audio is two-way again. The drawback is Asterisk cannot connects to the stun server after 7 to 8 hours. The strange one is another sip account has not one-way audio problem without stun server.
Firewall setting:
1. sip incoming through 5060.
2. rtp stream gets port range from 7000 to 7040. |