Hello Everyone, After a week of tinkering with opensips rtpproxy functions, I have a quite messy config file. Was wondering if anyone would be kind enough to share or walkthrough a configuration that will get two way audio working. Presently I have single outgoing audio. Seems like I am not able to pick up the callee's RTP.
INFO:remove_session: RTP stats: 0 in from callee, 872 in from caller, 872 relayed, 0 dropped INFO:remove_session: RTCP stats: 8 in from callee, 2 in from caller, 10 relayed, 0 dropped Basic layout of the network router 192.168.2.1 opensips 192.168.2.102 (bridged virutal box, ports forwarded) asterisk 192.168.2.110 (bridged virutal box) Polycom 192.168.2.11 [router]-----[opensips]-------[asterisk]--------[SIP Trunk] Please bare with the virtual box setup, I am just trying to get all the configs together before deploying onto the servers. I know i'm really close, and would love to be able to move on to the other parts (i.e., dialplan, routing etc...) I pasted an ngrep trace at http://pastebin.com/A39vBG3t. Thank you Kindly, Nick. _______________________________________________ Users mailing list [email protected] http://lists.opensips.org/cgi-bin/mailman/listinfo/users
