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opensips 基础代理设置

背景

最近梳理了opensips作为代理的配置,并结合rfc3665重新实现了该功能,现对其记录如下。

本次测试的opensips版本为:

version: opensips 3.5.5 (x86_64/linux)

实战

配置文件

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

xlog_level=4
#debug_mode=yes
stderror_enabled=yes
syslog_facility=LOG_LOCAL0

# Set up listeners
#socket=tls:127.0.0.1:5261
socket=udp:172.16.4.111:5260
socket=udp:172.16.4.111:5261

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

# set module path
mpath="/usr/local/lib64/sbc/modules/"


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

loadmodule "httpd.so"
loadmodule "mi_http.so"

modparam("httpd","port",9998)

#### 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@172.16.4.111/opensips")

#### REGISTRAR module
loadmodule "registrar.so"

#### RTPengine protocol
loadmodule "rtpengine.so"
modparam("rtpengine", "rtpengine_sock", "udp:172.16.4.111:2222")

#### Nathelper protocol
loadmodule "nathelper.so"
modparam("registrar|nathelper", "received_avp", "$avp(rcv)")

#### UDP protocol
loadmodule "proto_udp.so"
loadmodule "proto_tcp.so"

#### TLS protocol
#loadmodule "proto_tls.so"

#### WebSocket and WebSocketSecure protocol
#loadmodule "proto_wss.so"
loadmodule "proto_ws.so"



####### Routing Logic ########

# main request routing logic
route{
	if (!mf_process_maxfwd_header(10)) {
		sl_send_reply(483,"Too Many Hops");
		exit;
	}

	if (has_totag()) {
		# sequential requests within a dialog should
		# take the path determined by record-routing
		if (loose_route()) {
			if (is_method("INVITE")) {
				# even if in most of the cases is useless, do RR for
				# re-INVITEs alos, as some buggy clients do change route set
				# during the dialog.
				record_route();
			}

			xlog("L_DBG","[$cfg_line][$ci]---has-tag-:$rm|$rs|$tu|$socket_in(port)|$var(contact)\n");
			if ($si == "172.16.4.114") {
				$socket_out="udp:172.16.4.111:5261";
			} else {
				$socket_out="udp:172.16.4.111:5260";
			}
			# route it out to whatever destination was set by loose_route()
			# in $du (destination URI).
			route(relay);
		} else {
			if ( is_method("ACK") ) {
				if ( t_check_trans() ) {
					# non loose-route, but stateful ACK; must be an ACK after
					# a 487 or e.g. 404 from upstream server
					t_relay();
					exit;
				} else {
					# ACK without matching transaction ->
					# ignore and discard
					exit;
				}
			}
			sl_send_reply(404,"Not here");
		}
		exit;
	}

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

	t_check_trans();

	#if (!is_method("REGISTER")) {
	#	if (is_myself("$fd")) {
	#		# if caller is not local, then called number must be local
	#		if (!is_myself("$rd")) {
	#			send_reply(403,"Rely 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"))
			sl_send_reply(403,"Preload Route denied");
		exit;
	}

	# record routing
	if (!is_method("REGISTER|MESSAGE"))
		record_route_preset("172.16.4.111:5260");


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

	# requests for my domain
	if (is_method("PUBLISH|SUBSCRIBE")) {
		sl_send_reply(503, "Service Unavailable");
		exit;
	}

	# check if the clients are using WebSockets or WebSocketSecure
	if ( $socket_in(proto) == "WS"|| $socket_in(proto) == "WSS")
		setflag("SRC_WS");

	# consider the client is behind NAT - always fix the contact
	fix_nated_contact();
	xlog("L_DBG","[$cfg_line][$ci]---main-1-:$rm|$rs|$tu|$socket_in(port)|$var(contact)\n");

	if (is_method("REGISTER")) {

		# indicate that the client supports DTLS
		# so we know when he is called
		if (isflagset("SRC_WS"))
			setbflag("DST_WS");

		fix_nated_register();
		if (!save("location"))
			sl_reply_error();

		exit;
	}
	xlog("L_DBG","[$cfg_line][$ci]---main--:$rm|$rs|$tu|$socket_in(port)|$var(contact)\n");
	if ($rU==NULL) {
		# request with no Username in RURI
		sl_send_reply(484,"Address Incomplete");
		exit;
	}
	if ($si != "172.16.4.114") {
		$du = "sip:172.16.4.114:5060";
		$socket_out="udp:172.16.4.111:5260";
	}

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

	route(relay);
}

route[relay] {
	# for INVITEs enable some additional helper routes
	if (is_method("INVITE")) {
		t_on_branch("handle_nat");
		t_on_reply("handle_nat");
	} else if (is_method("BYE|CANCEL")) {
		rtpengine_delete();
	}

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

branch_route[handle_nat] {
	xlog("L_INFO","[$cfg_line][$ci]--m01---:$rm|$rs|$tu|$rU|$fU|$si|$socket_in(proto)|$socket_in(port)\n");
	if (!is_method("INVITE") || !has_body("application/sdp"))
		return;
	$var(rtp_flag) = "replace-origin ";

        if (isflagset("SRC_WS") && isbflagset("DST_WS")) { #web->web
            $var(rtp_flag) = $var(rtp_flag) + " ICE=force-relay DTLS=passive";
        } else if (isflagset("SRC_WS") && !isbflagset("DST_WS")){ #web->sip
            $var(rtp_flag) = $var(rtp_flag) + " codec-strip-G722 codec-strip-CN codec-strip-red strip-extmap rtcp-mux-demux DTLS=off SDES-off ICE=remove RTP/AVP";
        } else if (!isflagset("SRC_WS") && isbflagset("DST_WS")) {#sip->web
            $var(rtp_flag) = $var(rtp_flag) + " rtcp-mux-offer generate-mid DTLS=passive SDES-off ICE=force RTP/SAVPF";
        } else if (!isflagset("SRC_WS") && !isbflagset("DST_WS")) #sip->sip
            $var(rtp_flag) = $var(rtp_flag) + "  DTLS=off SDES-off ICE=remove RTP/AVP";

	rtpengine_offer("$var(rtp_flag)");
}

onreply_route[handle_nat] {

	xlog("L_INFO","[$cfg_line][$ci]--onreply-route---:$rm|$rs|$tu|$rU|$fU|$si|$socket_in(proto)|$socket_in(port)\n");
	if (is_present_hf("Record-Route")) {
		remove_hf("Record-Route");
		append_hf("Record-Route: <sip:172.16.4.111:5261;lr>\r\n");
		xlog("L_INFO","has record-route..\n");
	}
	fix_nated_contact();
	if (!has_body("application/sdp"))
		return;
	$var(rtp_flag) = "replace-origin ";
	if (isflagset("SRC_WS") && isbflagset("DST_WS")) #web->web
        $var(rtp_flag) = $var(rtp_flag) + " ICE=force-relay DTLS=passive";
    else if (isflagset("SRC_WS") && !isbflagset("DST_WS")) #web->sip
        $var(rtp_flag) = $var(rtp_flag) + " codec-strip-G722 codec-strip-CN codec-strip-red codec-strip-opus rtcp-mux-offer generate-mid DTLS=passive SDES-off ICE=force";
    else if (!isflagset("SRC_WS") && isbflagset("DST_WS")) #sip->web
        $var(rtp_flag) = $var(rtp_flag) + " rtcp-mux-offer generate-mid DTLS=passive SDES-off ICE=remove RTP/AVP";
    else if (!isflagset("SRC_WS") && !isbflagset("DST_WS")) #sip->sip
        $var(rtp_flag) = $var(rtp_flag) + "  DTLS=off SDES-off ICE=remove RTP/AVP";
	rtpengine_answer("$var(rtp_flag)");
}

测试

使用场景为:

场景

拨打的完整信令图:

sip

详细的信令信息:

  1. 拨打opensipsINVITE:

invite

主要看Request-URI,Via,Contact字段。

  1. opensips转发INVITE:

invite

这里我并没有修改Request-URI,添加了Record-Route.

这个Record-Route是为了uasBYE可以到正确的端口上。

Contactuas在接通之后,任何消息的Request-URI,比如本例中的INFO, BYE

之后的183200OK是通过Via找到回去的路由。

  1. opensips收到200OK后:

200ok

主要看Via字段。

  1. opensips转发200OK:

200ok

主要看Via,Record-Route,Contact字段。

这里的Record-Route是为了uas接通之后,发送信令如:BYE,INFO到正确的端口上.

Contactuac接通之后,任何消息的Request-URI,比如ACK,BYE

  1. opensips收到ACK:

ack

  1. opensips转发ACK:

ack

  1. opensips收到uasINFO:

info

  1. opensips转发INFO:

info

  1. opensips收到200OK:

200ok

  1. opensips转发200OK:

200ok

  1. opensips收到uacINFO:

info

  1. opensips转发INFO:

info

  1. opensips收到200ok:

200ok

  1. opensips转发200OK:

200ok

  1. opensips收到uasBYE:

bye

  1. opensips转发BYE:

bye

  1. opensips收到uac200OK:

200ok

  1. opensips转发200OK:

200ok

总结

  1. 当未接通时,需要设置Record-Route,$du,$socket_out

    前提是Contact未修改,如果Contact修改了,那么整个流程都要一起变。

  2. 当通话接通之后,整个流程都不需要设置,只设置$socket_out

问题: 可以看到uac在收到200OK时,Contactuas的地址,我们并不希望uac知道uas的地址。 uac后续的Request-URI都使用uas的地址,这个是不安全的, 需要修改Contactopensips的地址。

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