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.
------------------------------------------------------------------------------
October Webinars: Code for Performance
Free Intel webinars can help you accelerate application performance.
Explore tips for MPI, OpenMP, advanced profiling, and more. Get the most from
the latest Intel processors and coprocessors. See abstracts and register >
http://pubads.g.doubleclick.net/gampad/clk?id=60134791&iu=/4140/ostg.clktrk