Orlandinei Vujanski wrote:
> I have some users that connect via Softphone (SIP) outside my network.
> I've done a DNAT rule correctly.
> When these users connect, they can hear, but the other side can not hear.
> My telephony server receives connections by an alias eth0: 4 which is the 
> same IP output.

The "one way audio" SIP problem.
The problem is that SIP is one of those protocols that assumes a non-broken IP 
network, while NAT == Broken in IP terms. The aurdio is carried in an RTP 
stream, and the messages passed in teh SIP control packets reference the RTP 
endpoints by IP and Port. Of course, if one end says "Send your RTP stream to 
192.168.1.123 port 8000", and the other end is not in your private network, 
then packets to 192.168.1.123 get dropped.

So there are many appraoches to "fixing" this problem.

In the NAT gateway there may a SIP ALG (Application Level Gateway). This 
examines the SIP packets and mangles the contents to match - so it would change 
the 192.168.1.123 to whatever the public IP is. This may or may not work - and 
my preference is to turn this off.

Or the phone may use STUN to work out it's end - good luck if you are behind a 
Zyxel gateway, they suck big time.

Or the telephony server may handle it. In FreePBX there's an option against 
each extension for NAT - turning this on allows Asterisk to deal with the NAT 
at it's end and IME this works fairly well. My advice would be to look at your 
telephony server first and see what options it has for handling NAT.

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