[Serusers] Port Restricted Cone NAT

Hi all! Somone can give me an example how to forward calls to a (Cisco) PSTN Gateway with rtpproxy? The user is behind a Port Restricted Cone Nat and I have no idea how to solve the problem, that I have a one way audio stream. I'm using the following code: # special handling for NATed clients; first, nat test is # executed: it looks for via!=received and RFC1918 addresses # in Contact (may fail if line-folding used); also, # the received test should, if complete, should check all # vias for presence of received if (nat_uac_test("3")) { # allow RR-ed requests, as these may indicate that # a NAT-enabled proxy takes care of it; unless it is # a REGISTER if (method == "REGISTER" || ! search("^Record-Route:")) { log("LOG: Someone trying to register from private IP, rewriting\n"); # This will work only for user agents that support symmetric # communication. We tested quite many of them and majority is # smart smart enough to be symmetric. In some phones, like # it takes a configuration option. With Cisco 7960, it is # called NAT_Enable=Yes, with kphone it is called # "symmetric media" and "symmetric signaling". (The latter # not part of public released yet.) fix_nated_contact(); # Rewrite contact with source IP of signalling if (method == "INVITE") { fix_nated_sdp("1"); # Add direction=active to SDP }; force_rport(); # Add rport parameter to topmost Via setflag(6); # Mark as NATed }; }; [....] rewritehostport("xxx.xxx.xxx.xxx:5060"); #forward to the gateway [....] if (isflagset(6)) { force_rtp_proxy(); t_on_reply("2"); append_hf("P-Behind-NAT: Yes\r\n"); append_hf("P-hint: GATEWAY\r\n"); }; if (!t_relay()) { sl_reply_error(); break; }; [....] onreply_route[2] { if (status=~"2[0-9][0-9]") { force_rport(); fix_nated_contact(); fix_nated_sdp("2"); force_rtp_proxy(); }; } Has anyone of you experience with this? How you can imagine... I can't put the gateway on the ser proxy :-( Before the OK from the gateway reaches, the phone receives many UDP messages directly from the gateway... then the conversation is "switching". I made a ngrep on the messages to the PSTN and the media stream goes out like this... I'm wondering why the OK message has the IP information twice: INVITE: --------- v=0. o=- 3330963367 3330963367 IN IP4 192.168.1.101. s=SJphone. c=IN IP4 xxx.xxx.xxx.xxx. <-- ser server IP address t=0 0. a=direction:active. m=audio 35076 RTP/AVP 3 8 0 101. a=rtpmap:3 GSM/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-11,16. a=direction:active. a=nortpproxy:yes. OK (from GW to ser) ---------------------- v=0. o=CiscoSystemsSIP-GW-UserAgent 9220 7434 IN IP4 xxx.xxx.xxx.xxx. <-- IP of Gateway s=SIP Call. c=IN IP4 xxx.xxx.xxx.xxx. <-- IP of Gateway t=0 0. m=audio 16460 RTP/AVP 3 101. c=IN IP4 xxx.xxx.xxx.xxx. <-- IP of Gateway a=rtpmap:3 GSM/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-11,16. OK (from ser to phone): ------------------------ v=0. o=CiscoSystemsSIP-GW-UserAgent 7310 9703 IN IP4 xxx.xxx.xxx.xxx. <-- IP of the gateway s=SIP Call. c=IN IP4 xxx.xxx.xxx.xxx. <-- IP of us t=0 0. m=audio 35078 RTP/AVP 3 101. c=IN IP4 xxx.xxx.xxx.xxx. a=rtpmap:3 GSM/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-11,16. a=nortpproxy:yes. Thanks for your help! I hope there is a solution for this!! Sebastian