[OpenSER-Users] OpenSER as NAT traversal proxy HELP!
Joris Dobbelsteen
joris at familiedobbelsteen.nl
Tue Jul 22 02:25:15 CEST 2008
Neill Wilkinson wrote:
> If you are using IPtables and are familiar with how to add modules - there
> is a sip connection tracking module that might help:
>
> http://people.netfilter.org/chentschel/docs/sip-conntrack-nat.html
>
> Neill...;o)
>
> Neill Wilkinson
> Principal Consultant
>
> Aeonvista Ltd - opening up new ideas
>
I have that installed, but to the outside the SIP packets still carry
the LAN IP address. I'm currently missing audio (at least inbound is
nowhere to be seen) and it doesn't really work reliable at this moment.
That is a real problem currently and must be solved reliably.
The ZyXEL modem I have was intended to be the NAT router for the
network, but its configured differently in my case, so I can't make that
thing to play nicely with NAT.
lsmod on the firewall:
ip_nat_sip 8832 0
ip_conntrack_sip 13392 1 ip_nat_sip
Thanks so far,
- Joris
>
>
> -----Original Message-----
> From: users-bounces at lists.openser.org
> [mailto:users-bounces at lists.openser.org] On Behalf Of Joris Dobbelsteen
> Sent: 21 July 2008 21:10
> To: users at lists.openser.org
> Subject: [OpenSER-Users] OpenSER as NAT traversal proxy HELP!
>
> 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 at 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;
> }
>
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