Hi D'arcy Have you checked your RTP port ranges (I'm sure you have), and also that the server IP for RTP as specified in the initial SIP is correct?
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. 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. We subscribe to two trunk numbers provided by the VOIP service provider at each site in Asterisk. It turned out after carefully looking at the SIP flowing back and forth that the service provider was providing an RTP server IP that specified not the same IP as the SIP server (which is their standard practice) but a -different- RTP server IP. Due to the routing we have, neither system on either side of the SIP negotiated call could send packets to this "new" RTP server IP. We therefore added a route that specifically allowed that "new" RTP server IP to be reached by both machines on both sides of the VOIP service provider link. So can you carefully check that the SIP-negotiated RTP streams are going to IPs that are reachable in BOTH directions? 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. E. g. /etc/asterisk/rtp.conf specified 10000 - 50000 as allowable RTP ports, but my firewalld firewall under Centos was only allowing 10000 - 20000 - so I'd regularly get that my SECOND call to test the server would have audio in one direction - because Asterisk was allocating an RTP port on one side of the SIP call that was outside the range my firewalld was allowing. 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? Is that port allowed through all the PPP / network segments between you? Is the IP / IPs between you used to transfer RTP reachable from his side? Message: 1 Date: Tue, 11 Aug 2015 15:10:44 -0400 From: "D'Arcy J.M. Cain" <[email protected]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" <[email protected]> Subject: [asterisk-users] One way audio - doesn't seem to be NAT issue Message-ID: <20150811151044.79872ce9@imp> Content-Type: text/plain; charset=US-ASCII Given that both of us can make and accept calls and the server is simply connecting two separate channels I can't see where the problem might lie. Can anyone suggest a possible setup issue? I have tried so many things but I am willing to try them again. Feel free to make any suggestion no matter how silly. I really need to fix this. Cheers. -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
