I have a peculiar RTP issue. I'm experimenting with Jitsi as a softphone on one of my desktop Windows machines. That machine can either be connected to Asterisk via an VPN connection (with a static IP address) or not (via NAT). When it's connected via NAT, all is OK.
When it's connected with VPN, the following occurs: The voice path inbound to Jitsi works fine when Jitsi originates the call, no matter what it's calling. The voice path inbound to Jitsi works fine when it's called from another SIP device. The voice path inbound to Jitsi is silent when it's called from something on the other side of a PRI via DAHDI. I've run Wireshark on my desktop and see the RTP packets coming at the same rate and protocol (g711) in all the cases and "sip set debug peer xyz" doesn't shed any light on the situation (the SDP data looks similar in the working and non-worknig cases). Does anybody have any ideas what to look at next? -- _____________________________________________________________________ -- 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
