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

[D'Arcy J.M. Cain]

On Thu, 13 Aug 2015 10:41:31 +0200
"Stefan Viljoen" <viljoens at verishare.co.za> wrote:
> Have you checked your RTP port ranges (I'm sure you have), and also

Yes.  The ATA is using a range well within the range open on the server.

> 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.

> Not sure how this will relate to your setup, but we had something
> similar here using Asterisk 1.8.11.0 on both sides of the connection,
> via a VOIP service provider in the middle.

This is an Asterisk server talking to an ATA.

> 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?

> 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.

> It might require some careful tracing of SIP messages, maybe you can
> try this? Specifically try to determine what RTP port number is being
> negotiated when you have your zero-audio back from the remote party -
> what RTP port and RTP server IP is he using at that moment on his
> side?

I will check that.

Thanks for your suggestions.

-- 
D'Arcy J.M. Cain
System Administrator, Vex.Net
http://www.Vex.Net/ IM:darcy at Vex.Net
VoIP: sip:darcy at Vex.Net