[OpenSIPS-Users] Need some help with NAT/rtpproxy

James Lamanna jlamanna at gmail.com
Tue Dec 14 02:28:19 CET 2010


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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