Hi all, I was playing with openser 1.3 in a multihomed setup. In this setup, I noticed that by default, openser is mangling the Contact header and the sdp. I stripped down the config to bare minimum to see if it was something related to my config, but it seems that this is not related to the config. Here's the stripped down config that I used: # ----------- global configuration parameters ------------------------ debug=3 fork=yes log_stderror=no check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) auto_aliases=no listen=udp:br0 listen=udp:eth1 children=1 mhomed=1 disable_tcp=yes disable_dns_blacklist=true # ------------------ module loading ---------------------------------- mpath="/opt/lib/openser/modules" loadmodule "sl.so" loadmodule "tm.so" # -- tm params -- modparam("tm", "fr_inv_timer", 100) modparam("tm", "fr_inv_timer_avp", "$avp(i:26)") modparam("tm", "fr_timer", 10) modparam("tm", "fr_timer_avp", "$avp(i:25)") modparam("tm", "noisy_ctimer", 1) modparam("tm", "onreply_avp_mode", 1) loadmodule "rr.so" # ------------------------- request routing logic ------------------- # main routing logic route{ if (!method=="REGISTER") { record_route(); } if (loose_route()) { route(1); }; if (!uri==myself) { route(1); }; if (uri==myself) { sl_send_reply("404", "Not Found"); exit; }; route(1); } route[1] { if (!t_relay()) { sl_reply_error(); }; exit; } Here's the network setup: <asterisk box> --- < openser > --- <SIP server> <192.168.2.20> --- <192.168.2.1 : EXTERNAL_IP> --- <SIP_SERVER> And here's an ngrep of the initial INVITEs: 23:53:31.439678 IP 192.168.2.20.sip > 192.168.2.1.sip: SIP, length: 819 INVITE sip:callee@SIP_SERVER SIP/2.0 Via: SIP/2.0/UDP 192.168.2.20:5060;branch=z9hG4bK4f70ea70 From: "Ovi Sas" <sip:caller@SIP_SERVER>;tag=as048c5bed To: <sip:callee@SIP_SERVER> Contact: <sip:caller@192.168.2.20> Call-ID: 28aedce563bc57dc2cac78967e69c141@SIP_SERVER CSeq: 102 INVITE User-Agent: asterisk Max-Forwards: 70 Date: Thu, 18 Oct 2007 04:23:04 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 285 v=0 o=root 8560 8560 IN IP4 192.168.2.20 s=session c=IN IP4 192.168.2.20 t=0 0 m=audio 34028 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv 23:53:31.454727 IP EXTERNAL_IP.sip > SIP_SERVER.sip: SIP, length: 1011 INVITE sip:callee@SIP_SERVER SIP/2.0 Record-Route: <sip:EXTERNAL_IP;r2=on;lr;ftag=as048c5bed> Record-Route: <sip:192.168.2.1;r2=on;lr;ftag=as048c5bed> Via: SIP/2.0/UDP EXTERNAL_IP:5060;branch=z9hG4bKea4e.7aacc885.0 Via: SIP/2.0/UDP 192.168.2.20:5060;branch=z9hG4bK4f70ea70 From: "Ovi Sas" <sip:caller@SIP_SERVER>;tag=as048c5bed To: <sip:callee@SIP_SERVER> Contact: <sip:caller@EXTERNAL_IP:5060> Call-ID: 28aedce563bc57dc2cac78967e69c141@SIP_SERVER CSeq: 102 INVITE User-Agent: asterisk Max-Forwards: 70 Date: Thu, 18 Oct 2007 04:23:04 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 286 v=0 o=root 8560 8560 IN IP4 192.168.2.20 s=session c=IN IP4 EXTERNAL_IP t=0 0 m=audio 34028 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv Please note the Contact header and the c= line inside SDP. Is there a way to bypass this behaviour? Using the nathelper module doesn't seem to help (the Contact header will still carry the EXTERNAL_IP. Regards, Ovidiu Sas