比较Kamailio和OpenSIPS的重写contact函数

Kamailio：调用set_contact_alias()之后，在原有的contact的后面增加参数，具体地说，就是网络地址，网络端口和transport，好处是收到后续请求之时可以恢复原有contact的内容（当然也有坏处，就是增加参数之后导致包大，可能超过MTU）

OpenSIPS：调用fix_nated_contact()之后，用网络地址和网络端口直接覆盖了在原有的contact，那么当收到后续请求时不太可能恢复原有contact的内容。窃以为，一定要慎重

比如，下图所示，OpenSIPS就不要修改uac的contact，显然如此

UAC--->第三方Proxy--->OpenSIPS--->FreeSWITCH

参考链接：

https://kb.smartvox.co.uk/opensips/nat-contact-and-via-fixing-in-sip-part-3/

当然，我们可以模仿set_contact_alias函数

route[set_contact_alias] {
	if (is_present_hf("contact")) {
		$var(sut) = $si + "~" +  $sp + "~" + $socket_in(proto); # 远端网络地址
		$var(ct) = $(ct{re.subst,/^(.*)<(.*)>/\2/}); # 取contact <> 里面的内容
		# 更换成新的contact
		$var(new_ct) = "<" + $var(ct) + ";my_alias=" + $var(sut) + ">";
		remove_hf("contact");
		insert_hf("Contact: $var(new_ct)\r\n", "Call-ID");
	}
}

route[restore_contact_by_alias] {
	if (!has_totag()) { # 对话内请求才能调用这个路由
		return;
	}

	# xlog("***ru = $ru, du = $du\n");
	if ($du != NULL) { # 如果已经设置了$du，那么不能再调用这个路由
		return;
	}

	$var(my_alias) = $(ru{nameaddr.param,my_alias});
	# xlog("***my_alias = $var(my_alias)\n");

	if ($var(my_alias) != NULL) {
		$var(ip) = $(var(my_alias){s.select,0,~});
		$var(port) = $(var(my_alias){s.select,1,~});
		$var(transport) = $(var(my_alias){s.select,2,~});
		$var(url) = "sip:" + $var(ip) + ":" + $var(port) + ";transport=" + $var(transport);
		# xlog("url = $var(url)\n");
        setdsturi($var(url));
        $ru  = $(ru{re.subst,/^(sip:.*);my_alias=.*/\1/}); # 删除ru里面的my_alias以及内容		
	}
}

完整的路由脚本如下：

####### Global Parameters #########

/* uncomment the following lines to enable debugging */
#debug_mode=yes

log_level=3
xlog_level=3
stderror_enabled=no
syslog_enabled=yes
syslog_facility=LOG_LOCAL0

udp_workers=4

/* uncomment the next line to enable the auto temporary blacklisting of
   not available destinations (default disabled) */
#disable_dns_blacklist=no

/* uncomment the next line to enable IPv6 lookup after IPv4 dns
   lookup failures (default disabled) */
#dns_try_ipv6=yes

socket=udp:172.20.10.6:5060   # CUSTOMIZE ME
socket=tcp:172.20.10.6:5060   # CUSTOMIZE ME

####### Modules Section ########

#set module path
mpath="/usr/lib/x86_64-linux-gnu/opensips/modules/"

#### SIGNALING module
loadmodule "signaling.so"

#### StateLess module
loadmodule "sl.so"

#### Transaction Module
loadmodule "tm.so"
modparam("tm", "fr_timeout", 5)
modparam("tm", "fr_inv_timeout", 30)
modparam("tm", "restart_fr_on_each_reply", 0)
modparam("tm", "onreply_avp_mode", 1)

#### Record Route Module
loadmodule "rr.so"
/* do not append from tag to the RR (no need for this script) */
modparam("rr", "append_fromtag", 0)

#### MAX ForWarD module
loadmodule "maxfwd.so"

#### SIP MSG OPerationS module
loadmodule "sipmsgops.so"

#### FIFO Management Interface
loadmodule "mi_fifo.so"
modparam("mi_fifo", "fifo_name", "/tmp/opensips_fifo")
modparam("mi_fifo", "fifo_mode", 0666)

#### MYSQL module
loadmodule "db_mysql.so"

#### HTTPD module
loadmodule "httpd.so"
modparam("httpd", "port", 8888)

#### USeR LOCation module
loadmodule "usrloc.so"
modparam("usrloc", "nat_bflag", "NAT")
modparam("usrloc", "working_mode_preset", "single-instance-sql-write-back")
modparam("usrloc", "db_url",
	"mysql://opensips:opensipsrw@localhost/opensips") # CUSTOMIZE ME

#### REGISTRAR module
loadmodule "registrar.so"
modparam("registrar", "tcp_persistent_flag", "TCP_PERSISTENT")
modparam("registrar", "received_avp", "$avp(received_nh)")/* uncomment the next line not to allow more than 10 contacts per AOR */
#modparam("registrar", "max_contacts", 10)

#### ACCounting module
loadmodule "acc.so"
/* what special events should be accounted ? */
modparam("acc", "early_media", 0)
modparam("acc", "report_cancels", 0)
/* by default we do not adjust the direct of the sequential requests.
   if you enable this parameter, be sure to enable "append_fromtag"
   in "rr" module */
modparam("acc", "detect_direction", 0)

####  NAT modules
loadmodule "nathelper.so"
modparam("nathelper", "natping_interval", 10)
modparam("nathelper", "ping_nated_only", 1)
modparam("nathelper", "sipping_bflag", "SIP_PING_FLAG")
modparam("nathelper", "sipping_from", "sip:pinger@127.0.0.1") #CUSTOMIZE ME
modparam("nathelper", "received_avp", "$avp(received_nh)")

# loadmodule "rtpproxy.so"
# modparam("rtpproxy", "rtpproxy_sock", "udp:localhost:12221") # CUSTOMIZE ME

####  MI_HTTP module
loadmodule "mi_http.so"

loadmodule "proto_udp.so"
loadmodule "proto_tcp.so"
####### Routing Logic ########

# main request routing logic

route{
	xlog("$rm|$fU|$tU|$ci from $si:$sp\n");

	# initial NAT handling; detect if the request comes from behind a NAT
	# and apply contact fixing
	force_rport();
	#if (nat_uac_test(23)) {
	if (true) {
		if (is_method("REGISTER")) {
			fix_nated_register();
			setbflag("NAT");
		} else {
			# fix_nated_contact();
			route(set_contact_alias);
			setflag("NAT");
		}
	}

	if (!mf_process_maxfwd_header(10)) {
		send_reply(483,"Too Many Hops");
		exit;
	}

	if (has_totag()) {
		# handle hop-by-hop ACK (no routing required)
		if ( is_method("ACK") && t_check_trans() ) {
			t_relay();
			exit;
		}

		# sequential request within a dialog should
		# take the path determined by record-routing
		if ( !loose_route() ) {
			# we do record-routing for all our traffic, so we should not
			# receive any sequential requests without Route hdr.
			send_reply(404,"Not here");
			exit;
		}

		route(restore_contact_by_alias);
		if (is_method("BYE")) {
			# do accounting even if the transaction fails
			do_accounting("log","failed");
		}

		if (check_route_param("nat=yes"))
			setflag("NAT");
		# route it out to whatever destination was set by loose_route()
		# in $du (destination URI).
		route(relay);
		exit;
	}

	# CANCEL processing
	if (is_method("CANCEL")) {
		if (t_check_trans())
			t_relay();
		exit;
	}

	# absorb retransmissions, but do not create transaction
	t_check_trans();

	if ( !(is_method("REGISTER")  ) ) {
		if (is_myself("$fd")) {
		} else {
			# if caller is not local, then called number must be local
			if (!is_myself("$rd")) {
				send_reply(403,"Relay Forbidden");
				exit;
			}
		}

	}

	# preloaded route checking
	if (loose_route()) {
		xlog("L_ERR",
			"Attempt to route with preloaded Route's [$fu/$tu/$ru/$ci]");
		if (!is_method("ACK"))
			send_reply(403,"Preload Route denied");
		exit;
	}

	# record routing
	if (!is_method("REGISTER|MESSAGE"))
		record_route();

	# account only INVITEs
	if (is_method("INVITE")) {
		do_accounting("log");
	}

	if (!is_myself("$rd")) {
		append_hf("P-hint: outbound\r\n");
		route(relay);
	}

	# requests for my domain

	if (is_method("PUBLISH|SUBSCRIBE")) {
		send_reply(503, "Service Unavailable");
		exit;
	}

	if (is_method("REGISTER")) {
		if ($socket_in(proto) == "tcp")
			setflag("TCP_PERSISTENT");
		if (isflagset("NAT")) {
			setbflag("SIP_PING_FLAG");
		}
		# store the registration and generate a SIP reply
		if (!save("location"))
			xlog("failed to register AoR $tu\n");

		exit;
	}

	if ($rU==NULL) {
		# request with no Username in RURI
		send_reply(484,"Address Incomplete");
		exit;
	}

	# do lookup with method filtering
	if (!lookup("location", "method-filtering")) {
		t_reply(404, "Not Found");
		exit;
	}

	if (isbflagset("NAT")) setflag("NAT");

	# when routing via usrloc, log the missed calls also
	do_accounting("log","missed");
	route(relay);
}

route[relay] {
	# for INVITEs enable some additional helper routes
	if (is_method("INVITE")) {
		# if (isflagset("NAT") && has_body("application/sdp")) {
		#	rtpproxy_offer("ro");
		# }

		t_on_branch("per_branch_ops");
		t_on_reply("handle_nat");
		t_on_failure("missed_call");
	}

	if (isflagset("NAT")) {
		add_rr_param(";nat=yes");
	}

	if (!t_relay()) {
		send_reply(500,"Internal Error");
	}
	exit;
}

branch_route[per_branch_ops] {
	xlog("new branch at $ru\n");
}

onreply_route[handle_nat] {
	#if (nat_uac_test(1))
	#	fix_nated_contact();

	route(set_contact_alias);

	# if ( isflagset("NAT") && has_body("application/sdp") )
	#	rtpproxy_answer("ro");
	xlog("incoming reply\n");
}

failure_route[missed_call] {
	if (t_was_cancelled()) {
		exit;
	}

	# uncomment the following lines if you want to block client
	# redirect based on 3xx replies.
	##if (t_check_status("3[0-9][0-9]")) {
	##t_reply(404,"Not found");
	##	exit;
	##}
}

route[set_contact_alias] {
	if (is_present_hf("contact")) {
		$var(sut) = $si + "~" +  $sp + "~" + $socket_in(proto); # 远端网络地址
		$var(ct) = $(ct{re.subst,/^(.*)<(.*)>/\2/}); # 取contact <> 里面的内容
		# 更换成新的contact
		$var(new_ct) = "<" + $var(ct) + ";my_alias=" + $var(sut) + ">";
		remove_hf("contact");
		insert_hf("Contact: $var(new_ct)\r\n", "Call-ID");
	}
}

route[restore_contact_by_alias] {
	if (!has_totag()) { # 对话内请求才能调用这个路由
		return;
	}

	# xlog("***ru = $ru, du = $du\n");
	if ($du != NULL) { # 如果已经设置了$du，那么不能再调用这个路由
		return;
	}

	$var(my_alias) = $(ru{nameaddr.param,my_alias});
	# xlog("***my_alias = $var(my_alias)\n");

	if ($var(my_alias) != NULL) {
		$var(ip) = $(var(my_alias){s.select,0,~});
		$var(port) = $(var(my_alias){s.select,1,~});
		$var(transport) = $(var(my_alias){s.select,2,~});
		$var(url) = "sip:" + $var(ip) + ":" + $var(port) + ";transport=" + $var(transport);
		# xlog("url = $var(url)\n");
        setdsturi($var(url));
        $ru  = $(ru{re.subst,/^(sip:.*);my_alias=.*/\1/}); # 删除ru里面的my_alias以及内容		
	}
}

以上脚本在OpenSIPS3.4.3上调试通过，但没测试其他版本

于2024年2月8日（次日即除夕）

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