[asterisk-users] One way audio - doesn't seem to be NAT issue

[Stefan Viljoen]

Hi D'Arcy

>> that the server IP for RTP as specified in the initial SIP is correct?

>Both the server and client are outside of NAT so I don't know what this
might mean.  They both have public IPs.

This was a problem we had when the RTP server negotiated in SIP with our
VOIP ITSP on one side of the connection, differed from the IP we were
expecting on that side of the connection and was blocked in our firewall.

Once we perused the SIP traffic we noted this and added the extra IP to the
firewall for RTP traffic.

>> We had slightly different parameters, e. g. that we would have no RTP 
>> at all, but a call that did connect to total silence, dialed from 
>> either side.

>Was NAT involved?

Yes, NAT was being done at both ends, but it turned out that NATing was not
the problem.

>> Also check what RTP port ranges are being used - I have had this 
>> one-directional problem where the port range in /etc/asterisk/rtp.conf 
>> was too broad, and the firewall on my server was only allowing a 
>> smaller subset of RTP ports.

>rtpstart=10000
>rtpend=20000

>which is exactly what my packet filter allows through.

I assume you have tried turning your packet filter or firewall off
completely (just for a moment) to see if it helped?