[OpenSIPS-Users] Need some help with NAT/rtpproxy
James Lamanna
jlamanna at gmail.com
Tue Dec 14 02:54:23 CET 2010
Some other weird stuff that happens if I remove the fix_nated_register call:
Sometimes everything will work fine, audio is fine, etc..
However, sometimes I'll call a phone, and the phone will immediately
place a call back out to the number I'm dialing from!
(very strange).
-- James
On Mon, Dec 13, 2010 at 5:28 PM, James Lamanna <jlamanna at gmail.com> wrote:
> Hi,
> I'm having some issues getting a correct NAT configuration going.
> The problem I'm having is
> I get a "479 We don't forward to private IP addresses" back when
> receiving a call to a phone from Asterisk, presumably because the
> location table has private IPs in it for some reason.
>
> This seems to be related to my failed attempt to use fix_nated_register().
> Removing the call to fix_nated_register and just using
> fix_nated_contact allows calls to go through, but then I get no audio
> on either side...
>
> Config follows.
>
> Thanks.
>
> -- James
>
>
> debug=3 # debug level (cmd line: -dddddddddd)
> fork=yes
> log_stderror=no # (cmd line: -E)
> log_facility=LOG_LOCAL0
> tos=0x60
>
> # Uncomment these lines to enter debugging mode
> #fork=no
> #log_stderror=yes
> #debug=6
>
> check_via=no # (cmd. line: -v)
> dns=no # (cmd. line: -r)
> rev_dns=no # (cmd. line: -R)
> port=5060
> children=4
>
> listen=udp:208.xxx.xxx.6:5060
> listen=udp:208.xxx.xxx.6:5061
> # ------------------ module loading ----------------------------------
>
> #set module path
> #mpath="/usr/local/lib/opensips/modules/"
> mpath="/usr/local/lib64/opensips/modules/"
>
> # Uncomment this if you want to use SQL database
> loadmodule "db_mysql.so"
>
> loadmodule "sl.so"
> loadmodule "maxfwd.so"
> loadmodule "textops.so"
> loadmodule "avpops.so"
> loadmodule "tm.so"
> loadmodule "rr.so"
> loadmodule "dialog.so"
> loadmodule "signaling.so"
> loadmodule "options.so"
> loadmodule "localcache.so"
>
> loadmodule "usrloc.so"
>
> loadmodule "presence.so"
> loadmodule "presence_xml.so"
> loadmodule "presence_dialoginfo.so"
> loadmodule "pua.so"
> loadmodule "pua_dialoginfo.so"
> #loadmodule "pua_bla.so"
> loadmodule "pua_usrloc.so"
>
> loadmodule "registrar.so"
> loadmodule "mi_fifo.so"
> #loadmodule "xlog.so"
>
> # Uncomment this if you want digest authentication
> # db_mysql.so must be loaded !
> loadmodule "auth.so"
> loadmodule "auth_db.so"
>
> # !! Nathelper
> loadmodule "nathelper.so"
>
>
> # ----------------- setting module-specific parameters ---------------
>
> # -- mi_fifo params --
> modparam("mi_fifo", "fifo_name", "/tmp/opensips_fifo")
>
> modparam("usrloc", "db_mode", 2)
> modparam("usrloc|dialog|dispatcher|presence|presence_xml|pua|avpops",
> "db_url", "mysql://opensips:xxxxxxxxxxxxx@localhost/opensips")
>
>
> modparam("avpops","avp_table","usr_preferences")
>
> #modparam("dispatcher", "force_dst", 1)
> # Only use username
> #modparam("dispatcher", "flags", 1)
>
> # Store passwords for 1 hour in cache
>
> modparam("auth","username_spec","$avp(i:54)")
> modparam("auth","password_spec","$avp(i:55)")
> modparam("auth","calculate_ha1",1)
>
> modparam("auth_db", "db_url",
> "mysql://opensipsro:xxxxxxxx@localhost/opensips")
> modparam("auth_db", "calculate_ha1", yes)
> modparam("auth_db", "password_column", "password")
> modparam("auth_db", "load_credentials", "$avp(i:55)=password")
>
> modparam("rr", "enable_full_lr", 1)
>
> modparam("dialog", "dlg_flag", 4)
> modparam("dialog", "profiles_with_value", "caller")
>
> modparam("usrloc","nat_bflag",6)
> modparam("nathelper","sipping_bflag",8)
> #modparam("nathelper", "natping_interval", 30)
> modparam("nathelper", "ping_nated_only", 1) # Ping only clients behind NAT
> #modparam("nathelper", "natping_interval", 30)
> modparam("nathelper", "sipping_from", "sip:pinger at 208.xxx.xxx.6")
> modparam("nathelper", "rtpproxy_sock", "unix:/var/run/rtpproxy/rtpproxy.sock")
> modparam("nathelper", "received_avp", "$avp(i:42)")
> modparam("registrar", "received_avp", "$avp(i:42)")
>
> modparam("presence", "server_address", "sip:sa at 208.xxx.xxx.6:5060")
> modparam("presence", "expires_offset", 10)
> modparam("presence", "mix_dialog_presence", 1)
> #modparam("presence", "fallback2db", 1)
> modparam("presence_xml", "force_active", 1)
>
> modparam("presence_dialoginfo", "force_single_dialog", 1)
> modparam("pua_dialoginfo", "presence_server", "sip:sa at 208.xxx.xxx.6:5060")
> modparam("pua_dialoginfo", "include_callid", 1)
> modparam("pua_dialoginfo", "include_tags", 1)
> modparam("pua_dialoginfo", "caller_confirmed", 1)
>
> modparam("pua_usrloc", "default_domain", "208.xxx.xxx.6")
> modparam("pua_usrloc", "presence_server", "sip:sa at 208.xxx.xxx.6:5060")
>
> #modparam("stun","primary_ip","208.xxx.xxx.6")
> #modparam("stun","alternate_ip","208.90.184.10")
> #modparam("stun","primary_port","5060")
> #modparam("stun","alternate_port","3479")
>
> # ------------------------- request routing logic -------------------
>
> # main routing logic
>
> route{
>
> if (!is_method("NOTIFY"))
> xlog("L_INFO", "New request - Request/failure/branch routes: M=$rm
> RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n");
>
> # max_forwards==0, or excessively long requests
> if (!mf_process_maxfwd_header("10")) {
> sl_send_reply("483","Too Many Hops");
> exit;
> };
> if (msg:len >= 2048 ) {
> sl_send_reply("513", "Message too big");
> exit;
> };
>
> # !! Nathelper
> # Special handling for NATed clients; first, NAT test is
> # executed: it looks for via!=received and RFC1918 addresses
> # in Contact (may fail if line-folding is used); also,
> # the received test should, if completed, should check all
> # vias for rpesence of received
> if (nat_uac_test("19")) {
> # Allow RR-ed requests, as these may indicate that
> # a NAT-enabled proxy takes care of it; unless it is
> # a REGISTER
>
> if (is_method("REGISTER") || !is_present_hf("Record-Route")) {
> #xlog("L_INFO", "LOG:Someone trying to register from private IP,
> rewriting\n");
> #xlog("L_INFO", "$rb\n");
> # This will work only for user agents that support symmetric
> # communication. We tested quite many of them and majority is
> # smart enough to be symmetric. In some phones it takes a
> # configuration option. With Cisco 7960, it is called
> # NAT_Enable=Yes, with kphone it is called "symmetric media" and
> # "symmetric signalling".
>
> # Rewrite contact with source IP of signalling
> if (is_method("REGISTER")) {
> fix_nated_register();
> } else {
> fix_nated_contact();
> };
>
> if ( is_method("INVITE") ) {
> #xlog("L_INFO", "NAT: FIXING SDP");
> fix_nated_sdp("1"); # Add direction=active to SDP
> };
> force_rport(); # Add rport parameter to topmost Via
> setbflag(6); # Mark as NATed
>
> # if you want sip nat pinging
> # setbflag(8);
> };
> };
>
> # 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);
> exit;
> };
>
> # 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 (!is_method("REGISTER"))
> record_route();
>
> if (method == "INVITE") {
> setflag(4);
> }
>
> if (!uri==myself) {
> # mark routing logic in request
> append_hf("P-hint: outbound\r\n");
> route(1);
> exit;
> };
>
> # 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 (is_method("OPTIONS") && (! uri=~"sip:.*[@]+.*")) {
> options_reply();
> exit;
> }
>
> # if (is_method("INVITE|ACK")) {
> # unforce_rtp_proxy();
> # }
>
>
>
> if (is_method("REGISTER")) {
> #xlog("L_INFO", "trying to register $au $ad\n");
>
> if(cache_fetch("local","passwd_$tu",$avp(i:55))) {
> $avp(i:54) = $tU;
> xlog("SCRIPT: stored password is $avp(i:55)\n");
> # perform auth from variables
> # $avp(i:54) contains the username
> # $avp(i:55) contains the password
> if (!pv_www_authorize("asterisk")) {
> # authentication failed -> do challenge
> www_challenge("asterisk", "0");
> exit;
> };
> } else {
> # perform DB authentication ->
> # password will be loaded from DB automatically
> if (!www_authorize("asterisk", "subscriber")) {
> # authentication failed -> do challenge
> www_challenge("asterisk", "0");
> exit;
> };
> # after DB authentication, the password is available
> # in $avp(i:55) because of the "load_credentials"
> # module parameter.
> xlog("SCRIPT: storing password <$avp(i:55)>\n");
> # use a 20 minutes lifetime for the password;
> # after that, it will erased from cache and we do
> # db authentication again (refresh the passwd from DB)
> cache_store("local","passwd_$tu","$avp(i:55)",3600);
> }
>
> # Uncomment this if you want to use digest authentication
> #if (!www_authorize("asterisk", "subscriber")) {
> # www_challenge("asterisk", "0");
> # return;
> #};
>
> #bla_set_flag();
>
> save("location");
> pua_set_publish();
> exit;
> };
>
> lookup("aliases");
> if (!uri==myself) {
> append_hf("P-hint: outbound alias\r\n");
> route(1);
> exit;
> };
>
> #xlog("L_INFO", "TESTING FOR $hdr(Event)\n");
> if (is_method("NOTIFY") && $hdr(Event) == "message-summary") {
> #xlog("L_INFO", "MWI Notification $rb\n");
> if (!lookup("location")) {
> sl_send_reply("404", "Not Found");
> exit;
> }
> } else if (is_method("SUBSCRIBE") && (uri =~ "sip:[7-9][0-9]@208.xxx.xxx.6" ||
> $hdr(Event) == 'message-summary')) {
> xlog("L_INFO", "SUBSCRIBE FOR PAGE/VM \n");
> if(!cache_fetch("local","ast_$fU",$avp(i:200)))
> avp_db_load("$fu/username","$avp(i:200)");
> if ($avp(i:200) == NULL || $avp(i:200) == '') {
> xlog("INVALID DIALPLAN SERVER URL\n");
> sl_send_reply("404", "Not Found");
> exit;
> } else {
> cache_store("local","ast_$fU","$avp(i:200)",3600);
> }
> #rewritehostport("$avp(i:200)");
> $rd = $(avp(i:200){s.select,0,:});
> $rp = $(avp(i:200){s.select,1,:});
> } else if (is_method("PUBLISH|SUBSCRIBE|NOTIFY")) {
> route(2);
>
> # Asterisk signaling comes in on 5061
> } else if (dst_port==5061) {
> if (!lookup("location")) {
> sl_send_reply("404", "Not Found");
> exit;
> }
>
> if (is_method("INVITE")) {
> dialoginfo_set();
> }
>
>
> xlog("L_INFO", "request from asterisk $ru $tu\n");
> if (to_uri =~ ".*intercom=true") {
> xlog("INTERCOM REQUEST\n");
> $var(checkuser) = $tU;
>
> get_profile_size("caller","$var(checkuser)","$var(rcalls)");
> if ($var(rcalls) > 0) {
> xlog("DENY INTERCOM\n");
> sl_send_reply("486", "Busy Here");
> exit;
> }
> }
>
> if (!isflagset(31)) {
> #get_profile_size("caller","$avp(s:caller_uuid)","$var(calls)");
> create_dialog();
> #set_dlg_profile("caller","$avp(s:caller_uuid)");
> set_dlg_profile("caller","$tU");
> setflag(31);
> get_profile_size("caller","$tU","$var(calls)");
> xlog("NUM CALLS $tU $ru $mf $var(calls) \n");
> }
> } else if (is_method("INVITE")) {
> #if (!proxy_authorize("asterisk", "subscriber")) {
> # proxy_challenge("asterisk", "1"); # Realm will be autogenerated
> # exit;
> #};
>
> if(!cache_fetch("local","ast_$fU",$avp(i:200)))
> avp_db_load("$fu/username","$avp(i:200)");
> if ($avp(i:200) == NULL || $avp(i:200) == '') {
> xlog("INVALID DIALPLAN SERVER URL\n");
> sl_send_reply("404", "Not Found");
> exit;
> } else {
> cache_store("local","ast_$fU","$avp(i:200)",3600);
> }
> #rewritehostport("$avp(i:200)");
> $rd = $(avp(i:200){s.select,0,:});
> $rp = $(avp(i:200){s.select,1,:});
>
> if (!isflagset(31)) {
> #get_profile_size("caller","$avp(s:caller_uuid)","$var(calls)");
> create_dialog();
> #set_dlg_profile("caller","$avp(s:caller_uuid)");
> set_dlg_profile("caller","$fU");
> setflag(31);
> get_profile_size("caller","$fU","$var(calls)");
> xlog("NUM CALLS $fU $ru $mf $var(calls) \n");
> }
> dialoginfo_set();
> }
> };
> append_hf("P-hint: usrloc applied\r\n");
>
> route(1);
> }
>
> route[1]
> {
> # !! Nathelper
> if (uri=~"[@:](192\.168\.|10\.|172\.(1[6-9]|2[0-9]|3[0-1])\.)" &&
> !search("^Route:")){
> sl_send_reply("479", "We don't forward to private IP addresses");
> exit;
> };
>
> # if client or server know to be behind a NAT, enable relay
> if (isbflagset(6)) {
> force_rtp_proxy();
> };
>
> # NAT processing of replies; apply to all transactions (for example,
> # re-INVITEs from public to private UA are hard to identify as
> # NATed at the moment of request processing); look at replies
> t_on_reply("1");
>
> # send it out now; use stateful forwarding as it works reliably
> # even for UDP2TCP
> if (!t_relay()) {
> sl_reply_error();
> };
> }
>
> # !! Nathelper
> onreply_route[1] {
> # NATed transaction ?
> if (isbflagset(6) && status =~ "(183)|2[0-9][0-9]") {
> fix_nated_contact();
> force_rtp_proxy();
> # otherwise, is it a transaction behind a NAT and we did not
> # know at time of request processing ? (RFC1918 contacts)
> } else if (nat_uac_test("1")) {
> fix_nated_contact();
> } else if (status =~ "407") {
> xlog("AUTH unset profile $tu $fu\n");
> unset_dlg_profile("caller","$fU");
> }
> }
>
> route[2] {
> xlog("L_INFO", "NOTIFY SUBSCRIBE PUBLISH route\n");
> if (!t_newtran()) {
> sl_reply_error();
> exit;
> };
>
> if(is_method("PUBLISH")) {
> if ($hdr(Sender) != NULL)
> handle_publish("$hdr(Sender)");
> else
> handle_publish();
> }
> else if( is_method("SUBSCRIBE")) {
> xlog("L_INFO", "Handle Subscribe\n");
> handle_subscribe();
> }
> else if (is_method("NOTIFY")) {
> if ($hdr(Event) != "keep-alive") {
> # bla_handle_notify();
> pua_update_contact();
> }
> t_reply("200", "OK");
> }
>
> exit;
> }
>
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