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 :
21926
등록일 :
2017.09.06
08:15:25 (*.160.88.18)
엮인글 :
http://webs.co.kr/index.php?document_srl=3311814&act=trackback&key=b3b
게시글 주소 :
http://webs.co.kr/index.php?document_srl=3311814
List of Articles
번호 제목 글쓴이 조회 수 추천 수 날짜
51 SIPSorcery admin 44471   2014-03-18
 
50 Ekiga (formely known as GnomeMeeting) is an open source SoftPhone admin 43114   2014-03-12
 
49 telepresence: Open Source SIP Telepresence/MCU admin 183066   2014-03-12
 
48 SIP PBX - OpenSIPS and Asterisk configuration admin 164293   2014-03-12
 
47 Conference Support in Kamailio (OpenSER) admin 86841   2014-03-12
 
46 OpenSIPS configuration for 2 or more FreeSWITCH installs admin 75412   2014-03-12
 
45 The Impact of TLS on SIP Server Performance file admin 42465   2014-03-12
 
44 book-opensips-101 / content / 3.2. SIP TLS Secure Calling.mediawiki admin 42981   2014-03-12
 
43 Where to check OpenSIPS does not start? admin 43459   2014-03-09
 
42 opensips-1.10.0_src.tar.gz experimental source code documentation admin 38314   2014-03-09
 
41 Kamailo OpenSIPs installation on Debian admin 83610   2014-03-09
 
40 Using the openSIPS Registrant Module admin 52830   2014-03-09
 
39 RTPproxy Frequentry Asked Questions (FAQ) ¶ admin 176524   2014-03-07
 
38 Building Telephony Systems with OpenSIPS 1.6 RTPProxy + OpenSIPS 1.7 admin 40915   2014-03-07
 
37 Installing RTPproxy Start RTPproxy in Bridged mode very good admin 103781   2014-03-07
 
36 OpenSIPS Control Panel (OCP) Installation Guide admin 281050   2014-03-06
 
35 OpenSIPS Control Panel install guide admin 96974   2014-03-06
 
34 rtpproxy Module admin 39111   2014-03-06
 
33 MediaProxy Installation Guide admin 182429   2014-03-06
 
32 How to install OpenSIPS on CentOS debian module add xcap admin 46356   2014-03-06
 
31 Problem with presence_xml module Opensips 1.9 admin 48323   2014-03-06
 
30 Building Telephony Systems with OpenSIPS 1.6 books file admin 42973   2014-03-06
 
29 Multimedia Service Platform admin 37719   2014-03-06
 
28 How to install OpenSIPS on CentOS Debian etc admin 45437   2014-03-05
 
27 Opensips Installation, How to. admin 76374   2014-03-05
 
26 100% CPU usage opensips admin 54944   2014-03-05
 
25 A2Billing and OpenSIPS admin 42358   2014-03-04
 
24 Opensips_1.9 install guide this is great I like this admin 108623   2014-03-04
 
23 Opensips install debian admin 38717   2014-03-03
 
22 Open Source VOIP applications, both clients and servers. admin 44822   2013-11-20