Dear Kamailio'ns,

I am working on Kamailio server (V 4.1.2) with RTPproxy (1.2.1) integrated, in a standalone intranet infrastructure (no any connection with internet). I dont have any NAT settings in my network set-up. even though i will not get Audio/video calls through some times. So in that concern I have installed RTPproxy, now all the audio/Video calls are fine (with some Pixelled). I have the Following Kamailio configuration script, in which it suppose to invoke RTPproxy service when the SIP clients behind NAT. But every time when i do Audio/Video calls, they are proxying through RTPproxy server only. I analysed SIP captures of Audio/video call, i didnt found any IP/port changes in the whole SIP session and with this i assumed that there is no NAT issue in my Network. But why all the Audio/Video calls are proxying through RTPproxy everytime ?

Is there any Wrong placement of function call in Kamailio configuration script (below) ? #----------------------------------------------------------------- #!ifdef WITH_NAT # ----- rtpproxy params ----- modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722")

# ----- nathelper params ----- modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", 7) modparam("nathelper", "sipping_from", "sip:pinger@192.168.2.52") modparam("nathelper", "sipping_method", "INFO")

# ----- NAT_traversal ----- modparam("nat_traversal", "keepalive_interval", 60) modparam("nat_traversal", "keepalive_method", "NOTIFY") modparam("nat_traversal", "keepalive_state_file", "/var/run/kamailio/keepalive_state")

# ----- params needed for NAT traversal in other modules ----- modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") modparam("usrloc", "nat_bflag", 6) #!endif

#Routing Script # ----------------------------------------------------------------- # Sanity Check Section # ----------------------------------------------------------------- route { if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); break; }; #if (msg:len > max_len) { if (msg:len >= 8192 ) { sl_send_reply("513", "Message too big"); break; };

# ----------------------------------------------------------------- # Record Route Section # ----------------------------------------------------------------- #if (method!="REGISTER") { if (!method=="REGISTER") { record_route(); }; if (method=="BYE" || method=="CANCEL") { # unforce_rtp_proxy(); rtpproxy_answer(); }

# ----------------------------------------------------------------- # Loose Route Section # ----------------------------------------------------------------- if (loose_route()) {

if ((method=="INVITE" || method=="REFER") && !has_totag()) { sl_send_reply("403", "Forbidden"); break; };

if (method=="INVITE") { if (!proxy_authorize("192.168.2.52","subscriber")) { # proxy_challenge("","0"); proxy_challenge("192.168.2.52", "0"); # break; }

else if (!check_from()) { sl_send_reply("403", "Use From=ID"); break; };

consume_credentials(); if (nat_uac_test("19")) { setflag(6); force_rport(); fix_nated_contact(); }; rtpproxy_offer("l"); }; route(1); break; };

# ----------------------------------------------------------------- # Call Type Processing Section # ----------------------------------------------------------------- # if (uri!=myself) { if (!uri==myself) { route(4);

route(1); break; };

if (method=="ACK") { route(1); break; }

if (method=="CANCEL") { route(1); break; }

else if (method=="INVITE") { route(3); break; } else if (method=="REGISTER") { route(2); break; };

lookup("aliases"); if (uri!=myself) { route(4);

route(1); break; };

if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; route(1); }

# ----------------------------------------------------------------- # Default Message Handler # ----------------------------------------------------------------- route[1] { t_on_reply("1");

if (!t_relay()) {

if (method=="INVITE" && isflagset(6)) { rtpproxy_answer();; };

sl_reply_error(); }; }

# ----------------------------------------------------------------- # REGISTER Message Handler # ----------------------------------------------------------------- route[2] { if (!search("^Contact:[ ]**") && nat_uac_test("19")) { setflag(6); fix_nated_register(); force_rport(); }; sl_send_reply("100", "Trying");

if (!www_authorize("192.168.2.52","subscriber")) { www_challenge("192.168.2.52","0"); break; };

if (!check_to()) { sl_send_reply("401", "Unauthorized"); break; };

consume_credentials(); if (!save("location")) { sl_reply_error(); }; }

# ----------------------------------------------------------------- # INVITE Message Handler # ----------------------------------------------------------------- route[3] { if (!proxy_authorize("192.168.2.52","subscriber")) { proxy_challenge("192.168.2.52","0"); break; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); break; }; consume_credentials(); if (nat_uac_test("19")) { setflag(6); }

lookup("aliases"); if (uri!=myself) { route(4);

route(1); break; };

if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; };

route(4); route(1); }

# ----------------------------------------------------------------- # NAT Traversal Section # ----------------------------------------------------------------- route[4] { if (isflagset(6)) { force_rport(); fix_nated_contact(); # force_rtp_proxy(); rtpproxy_offer(); } }

onreply_route[1] { if (isflagset(6) && status=~"(180)|(183)|2[0-9][0-9]") { if (!search("^Content-Length:[ ]*0")) { # force_rtp_proxy(); rtpproxy_offer(); }; };

if (nat_uac_test("1")) { fix_nated_contact(); }; }

Please find the Attachment for Tcpdump based Video call Sip capture for your better Understanding.

PS: Both Kamailio and RTPproxy are running on same IP (host), i.e 192.168.2.52.

Please anybody help me in resolving this issue.

Any help will greatly appreciate.

Regards, Ravi.