New subject: [Kamailio-Users] [OpenSER-Users] OpenSER as NAT traversal proxy HELP!

21 Jul 2008


      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=~"@tls_domain1.net") {
                 #       t_relay("tls:domain1.net");
                 #       exit;
                 #} else if(uri=~"@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;
}