VoIP：Asterisk/Freepbx-No audio for established sip calls and call drops after 30 seconds 两条命令解决Asterisk通话无声音、30秒自动挂断的问题

Environment:

Asterisk 13.15.1: NAT enabled, SIP ports opened in firewall.

Issues:

1. calls can be established but no audio on both sides
2. calls auto drop after 30 seconds

Reasons:

1. firewall stopped RTP packets as RTP ports not opened: audio transmission needs UDP ports (10000-20000) opened;
2. call will be cut if no RTP packets received in 30 seconds(this timeout can be customized in Chan SIP settings).

Solution:

I Googled a lot and didn't get any clear guides or posts providing a solution. I solved it by adding a firewall rule when I found asterisk calls are over RTP ports(range is 10000-20000).

#firewall-cmd --zone=public --add-port=10000-20000/udp --permanent
#firewall-cmd --reload

Good luck!