2 Apr
2014
2 Apr
'14
7:50 a.m.
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.