한국어

소프트스위치

온누리070 플레이스토어 다운로드
    acrobits softphone
     온누리 070 카카오 프러스 친구추가온누리 070 카카오 프러스 친구추가친추
     카카오톡 채팅 상담 카카오톡 채팅 상담카톡
    
     라인상담
     라인으로 공유

     페북공유

   ◎위챗 : speedseoul


  
     PAYPAL
     
     PRICE
     

pixel.gif

    before pay call 0088 from app




3.  Configuration file

The following configuration file is a minimal working example of a Residential script that can handle clients connections over both UDP and Websocket transports. This example assumes that the SDP offer is present in the INVITE from the UAC and the SDP answer is in the 200 OK from the UAS.

#
# OpenSIPS residential configuration script
#     by OpenSIPS Solutions <team@opensips-solutions.com>
#
# Please refer to the Core CookBook at:
#      http://www.opensips.org/Resources/DocsCookbooks
# for a explanation of possible statements, functions and parameters.
#


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

debug=3
log_stderror=no
log_facility=LOG_LOCAL0

fork=yes
children=4
auto_aliases=no

listen=udp:127.0.0.0:5060 # TODO: update with your local IP and port
listen=ws:127.0.0.0:8080 # TODO: update with your local IP and port

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

# set module path
mpath="/usr/local/lib/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"
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)

#### URI module
loadmodule "uri.so"
modparam("uri", "use_uri_table", 0)

#### USeR LOCation module
loadmodule "usrloc.so"
modparam("usrloc", "nat_bflag", "NAT")
modparam("usrloc", "db_mode",   0)

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

#### RTPengine protocol
loadmodule "rtpengine.so"
modparam("rtpengine", "rtpengine_sock", "udp:127.0.0.0:60000")

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

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

#### WebSocket protocol
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();
			}

			# 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 (from_uri!=myself) {
			# if caller is not local, then called number must be local
			if (!uri==myself) {
				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();

	if (!uri==myself) {
		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
	if (proto == WS)
		setflag(SRC_WS);

	# consider the client is behind NAT - always fix the contact
	fix_nated_contact();

	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;
	}

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

	# do lookup with method filtering
	if (!lookup("location","m")) {
		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] {

	if (!is_method("INVITE") || !has_body("application/sdp"))
		return;

	if (isflagset(SRC_WS) && isbflagset(DST_WS))
		$var(rtpengine_flags) = "ICE=force-relay DTLS=passive";
	else if (isflagset(SRC_WS) && !isbflagset(DST_WS))
		$var(rtpengine_flags) = "RTP/AVP replace-session-connection replace-origin ICE=remove";
	else if (!isflagset(SRC_WS) && isbflagset(DST_WS))
		$var(rtpengine_flags) = "UDP/TLS/RTP/SAVPF ICE=force";
	else if (!isflagset(SRC_WS) && !isbflagset(DST_WS))
		$var(rtpengine_flags) = "RTP/AVP replace-session-connection replace-origin ICE=remove";

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

onreply_route[handle_nat] {

	fix_nated_contact();
	if (!has_body("application/sdp"))
		return;

	if (isflagset(SRC_WS) && isbflagset(DST_WS))
		$var(rtpengine_flags) = "ICE=force-relay DTLS=passive";
	else if (isflagset(SRC_WS) && !isbflagset(DST_WS))
		$var(rtpengine_flags) = "UDP/TLS/RTP/SAVPF ICE=force";
	else if (!isflagset(SRC_WS) && isbflagset(DST_WS))
		$var(rtpengine_flags) = "RTP/AVP replace-session-connection replace-origin ICE=remove";
	else if (!isflagset(SRC_WS) && !isbflagset(DST_WS))
		$var(rtpengine_flags) = "RTP/AVP replace-session-connection replace-origin ICE=remove";

	rtpengine_answer("$var(rtpengine_flags)");
}
조회 수 :
22891
등록일 :
2017.09.06
08:15:25 (*.160.88.18)
엮인글 :
http://webs.co.kr/index.php?document_srl=3311814&act=trackback&key=202
게시글 주소 :
http://webs.co.kr/index.php?document_srl=3311814
List of Articles
번호 제목 글쓴이 조회 수 추천 수sort 날짜
22 OpenSIPS vs Asterisk admin 224634   2013-04-06
 
21 OpenSER_from_an_asterisk_POV file admin 40027   2013-04-06
 
20 rfc5766-turn-server admin 42413   2013-03-21
 
19 OfficeSIP Server is freeware VoIP, SIP server for Windows admin 56282   2013-09-11
 
18 Open Source VOIP applications, both clients and servers. admin 46520   2013-11-20
 
17 opensips Rtpengine installation configuration admin 167   2024-05-22
 
16 Asterisk 에서 FreeSWITCH 전환 admin 2882   2024-04-09
 
15 OpenSIPS 이해 서비스 지원 admin 2888   2024-04-09
 
14 OpenSIPS install configuration support admin 2921   2024-04-09
 
13 What is new in 1.8.0 opensip admin 257178   2012-05-21
 
12 opensips-cli command admin 9914   2023-08-07
 
11 string trans opensips admin 8093   2023-08-05
 
10 opensips Push Notification configuration admin 8206   2023-07-29
 
9 opensips Call pickup configuration admin 8114   2023-07-27
 
8 t_relay opensips admin 8021   2023-07-25
 
7 debian 11 opensips 3.2 install command admin 12362   2023-06-25
 
6 Asterisk v1.4x built on FreeBSD v7.1 UNIX admin 153823   2012-01-06
 
5 SIP 트래픽 생성 테스트 툴 admin 139883   2011-12-23
 
4 사설 망 환경에서 SIP 의 NAT Traversal 문제 admin 148428   2011-12-23
 
3 the OpenSIPS Project OpenSIP admin 46972   2011-12-14
 
2 OpenH323 Gatekeeper - The GNU Gatekeeper admin 54918   2011-12-14
 
1 The FreeRADIUS Project admin 45381   2011-12-14