Dear,
I'm really trying to use OpenSER as a NAT traversal SIP proxy, since my
home phone keeps breaking voice channels (the box was not intended
behind NAT and I'm, of course, using a configuration that no so well
supported).
What is the idea:
SIP transactions should travel this way:
ZyXEL UA <-> SIP Proxy <-> NAT Firewall (iptables) <-> {Internet}
RTP should travel this way:
ZyXEL UA <-> NAT Firewall & RTPProxy <-> {Internet}
My current test is using X-Lite with voipbuster, but that doesn't really
work. It seems that registers are functioning, at least X-Lite reports
itself being registered.
Voice calls always end up in timeouts, so something is really going
wrong here, it might be authentication problems?
An added problem is that I have just sufficient knowledge of SIP to see
what it is doing, without really knowing what to expect exactly.
Furthermore I have virtually no knowledge of OpenSER. I've quite a hard
time even grasping the configuration I typed in. This is not really helpful
What I do know:
* SIP Proxy traffic is flowing.
* SIP INVITES don't work at all.
* SIP to RTP is communication, but I don't know if RTP is actually flowing.
I stole most of the configuration from the "04 NAT Traversal" slides of
the "Italy 2007 Admin course", to which there is link on the
documentation site. I adapted it to make it work with the debian
supplied OpenSER 1.1.
How do I get this all working?
What am I getting wrong?
I really really appeciate any help I can get to get it working!
- Joris
Config is this:
# ----------- global configuration parameters ------------------------
debug=4 # debug level (cmd line: -dddddddddd)
fork=yes # Set to no to enter debugging mode
log_stderror=no # (cmd line: -E) Set to yes to enter debugging mode
check_via=no # (cmd. line: -v)
dns=no # (cmd. line: -r)
rev_dns=no # (cmd. line: -R)
advertised_address="82.168.191.xx"
advertised_port=5060
port=5060
children=4
fifo="/tmp/openser_fifo"
#
# ------------------ module loading ----------------------------------
# Uncomment this if you want to use SQL database
mpath="/usr/lib/openser/modules/"
loadmodule "mysql.so"
loadmodule "sl.so"
loadmodule "tm.so"
loadmodule "rr.so"
loadmodule "maxfwd.so"
loadmodule "usrloc.so"
loadmodule "registrar.so"
loadmodule "textops.so"
loadmodule "nathelper.so"
# Uncomment this if you want digest authentication
# mysql.so must be loaded !
loadmodule "auth.so"
loadmodule "auth_db.so"
# ----------------- setting module-specific parameters ---------------
# -- usrloc params --
modparam("usrloc", "db_mode", 0)
# Uncomment this if you want to use SQL database
# for persistent storage and comment the previous line
#modparam("usrloc", "db_mode", 2)
# -- auth params --
# Uncomment if you are using auth module
#
modparam("auth_db", "calculate_ha1", yes)
#
# If you set "calculate_ha1" parameter to yes (which true in this config),
# uncomment also the following parameter)
#
modparam("auth_db", "password_column", "password")
# -- rr params --
# add value to ;lr param to make some broken UAs happy
modparam("rr", "enable_full_lr", 1)
# -- nathelper params ---
modparam("nathelper", "rtpproxy_sock",
"udp:192.168.10.6:22222")
modparam("nathelper", "natping_interval", 30)
modparam("nathelper", "ping_nated_only", 1)
#modparam("nathelper", "sipping_bflag", 7)
modparam("nathelper", "sipping_from",
"sip:pinger@82.168.191.xx")
# ------------------------- request routing logic -------------------
# main routing logic
route{
# initial sanity checks -- messages with
# max_forwards==0, or excessively long requests
if (!mf_process_maxfwd_header("10")) {
sl_send_reply("483","Too Many Hops");
exit;
};
if (msg:len >= 2048 ) {
sl_send_reply("513", "Message too big");
exit;
};
# NAT detection
route(2);
# we record-route all messages -- to make sure that
# subsequent messages will go through our proxy; that's
# particularly good if upstream and downstream entities
# use different transport protocol
if (!method=="REGISTER")
record_route();
# subsequent messages withing a dialog should take the
# path determined by record-routing
if (loose_route()) {
# mark routing logic in request
append_hf("P-hint: rr-enforced\r\n");
route(1);
};
if (!uri==myself) {
# mark routing logic in request
append_hf("P-hint: outbound\r\n");
# if you have some interdomain connections via TLS
#if(uri=~"(a)tls_domain1.net") {
# t_relay("tls:domain1.net");
# exit;
#} else if(uri=~"(a)tls_domain2.net") {
# t_relay("tls:domain2.net");
# exit;
#}
route(1);
};
# if the request is for other domain use UsrLoc
# (in case, it does not work, use the following command
# with proper names and addresses in it)
if (uri==myself) {
if (method=="REGISTER") {
# Uncomment this if you want to use digest
authentication
if (!www_authorize("sip.familiedobbelsteen.nl",
"subscriber")) {
www_challenge("sip.familiedobbelsteen.nl", "0");
exit;
};
if (isflagset(5)) {
# set branch flag -- when someone will
call this user
# INVITE will have branch flag 6 set
after loopup("location")
setflag(6);
# if you want OPTIONS natpings
uncomment next
# setflag(7);
};
save("location");
exit;
};
lookup("aliases");
if (!uri==myself) {
append_hf("P-hint: outbound alias\r\n");
route(1);
};
# native SIP destinations are handled using our USRLOC DB
if (!lookup("location")) {
sl_send_reply("404", "Not Found");
exit;
};
append_hf("P-hint: usrloc applied\r\n");
};
route(1);
}
route[1] {
# send it out now; use stateful forwarding as it works reliably
# even for UDP2TCP
if (subst_uri('/(sip:.*);nat=yes/\1/i')) {
setflag(6);
};
if (isflagset(5) || isflagset(6)) {
route(3);
};
if (!t_relay()) {
sl_reply_error();
};
exit;
}
route[2] {
force_rport();
if(nat_uac_test("19")) {
if (method=="REGISTER") {
fix_nated_register();
} else {
fix_nated_contact();
};
setflag(5);
};
}
route[3] {
if (is_method("BYE")) {
unforce_rtp_proxy();
} else if (is_method("INVITE")) {
force_rtp_proxy("", "82.168.191.xx");
t_on_failure("2");
};
if (isflagset(5))
search_append('Contact:.*sip:[^>[:cntrl:]]*',
';nat=yes');
t_on_reply("1");
}
failure_route[2] {
if (isflagset(6)||isflagset(5)) {
unforce_rtp_proxy();
};
}
onreply_route[1] {
if ((isflagset(5) || isflagset(6)) && status =~
"(183)|(2[0-9][0-9])") {
force_rtp_proxy();
};
search_append('Contact:.*sip:[^>[:cntrl:]]*', ';nat=yes');
if (isflagset(6)) {
fix_nated_contact();
};
exit;
}