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One way audio problem

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.

Have set all the extension using nat=yes even though they are located behind the NAT including your firewall?

YH

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Will ports >10000 be better for rtp streams?

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Speaking of ports, I would like to share the list of ports assignment as attached.

Port Numbers - IANA — Internet Assigned Numbers Authority.rar (207.09 KB)

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Actually, the port number is not so importance as rtp.conf and the firewall have the same ports.

The wired stuff is another VSP does not experience one-way audio while the one does. Both two have the same settings. I suspect that it should relate to the NAT implementation of VSP.

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The warning message is gone when deaktivate the linux tracking modules
http://www.voip-info.org/wiki/view/ip_nat_sip

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