소프트스위치

오늘:
2,206
어제:
2,557
전체:
2,715,055

고객센타 : 070-7752-2000
팩스 : 070-7752-2001
휴대폰 : 010-9513-0019
email : voipkorea@yahoo.co.kr

국민은행
(주)제이에스솔루션
047101-04-155519

Flag Counter
■ 무료 : 유선 집전화 휴대폰 ( 한국 미국 중국 카나다) ↔ (국내 해외 여행자 상사 주재원 유학생) / 가입무 무제한무료■


http://www.opensips.org/Documentation/Tutorials-WebSocket-2-1



1.  Tutorial Overview

WebSocket is a protocol that provides full-duplex communication between web clients and servers over TCP connections. Using the WebSocket protocol, browsers can connect to web servers and exchange data, regardless the type or nature of the application protocol. RFC 7118 leveraged this protocol in order to allow browsers to make VoIP calls using the SIP protocol.

This document describes how to use OpenSIPS as the core component of a SIP platform that connects both SIP clients (over UDP, TCP or TLS) as well as browser based clients (using SIP over WebSockets). While OpenSIPS handles the SIP signalling part, media is handled by RTPengine, a high performance media proxy that is able to handle both RTP and SRTP media streams, as well as bridging between them.

This tutorial is inspired from

2.  Setup

2.1  RTPengine

Installation

The RTPengine consists of two main components: a kernel module used to efficiently route the RTP packets directly in kernel, and a daemon used to communicate with OpenSIPS. You can find more details here. Both components can be installed from debs (on Debian based systems) or directly from sources. Simply follow the official documentation to install RTPengine.

Usage

After installing the kernel module and the additional libraries, the rtpengine daemon has to be configured. This can be done from /etc/default/ngcp-rtpengine-daemon if installed from debs, or from the command line if the daemon is started manually. On systemd based OSes, Eric Tamme created some startup scripts.

The interesting parameters we are using are as follows:

  • -i: the listening interface for RTP/SRTP
  • -n: the listening IP and port that is used by OpenSIPS to communicate with the RTPengine (NOTE: the rtpengine module only works with the rtpengine NG protocol, so you must use -n/--listen-ng; Using -u/--listen-udp or -l/--listen-tcp will not work!)
  • -c: the IP and port of the CLI - this is used to gather statistics for the RTP/SRTP sessions
  • -m, -M: both take an integer as argument and together define the local port range from which rtpengine will allocate UDP ports for media traffic relay. Default to 30000 and 40000 respectively.
  • -L: indicates the debugging level

You can find all the parameters available here.

Here is an example that runs rtpengine from cli that talks with OpenSIPS over localhost and RTP using the 1.1.1.1 IP:

./rtpengine -p /var/run/rtpengine.pid -i eth0/1.1.1.1 -n 127.0.0.1:60000 -c 127.0.0.1:60001 -m 50000 -M 55000 -E -L 7
Troubleshoot

First make sure the rtpengine daemon is started:

ps -ef | grep rtpengine

If the rtpengine daemon does not start, make sure the xt_RTPENGINE kernel module is loaded:

lsmod | grep xt_RTPENGINE

If the module is not loaded, make sure the ip_tables and x_tables kernel modules are loaded. Also, check the logs for the last errors of the system

dmesg

2.2  OpenSIPS

In order to use WebSocket in OpenSIPS, one has to load the proto_ws into its configuration file and define a listener for the WebSocket protocol.

listen=ws:127.0.0.1:8080
...
loadmodule "proto_ws.so"

Next, the rtpengine module has to be loaded and configured to communicate with the rtpengine daemon.

loadmodule "rtpengine.so"
modparam("rtpengine", "rtpengine_sock", "udp:127.0.0.1:60000")

Note that the rtpengine_sock parameter should be the same as the -n parameter sent to the rtpengine daemon, and OpenSIPS should have IP connectivity to that socket.

Next, the routing logic has to be changed in order to treat different the clients that use DTLS-SRTP, from the ones that use plain RTP and enable bridging if necessary. To do that, one can check if the request message was received over the WebSocket protocol. This can be achieved using the following code:

if (proto == WS)
    setflag(SRC_WS);

In case the request is a REGISTER, we want to store this information in the location table, so that we know then the user is called. To do that, we can set a branch flag before calling the save()function. This way, when the lookup() method returns, we will be able to determine whether the client uses WebSocket or not.

    if (is_method("REGISTER")) {
        if (isflagset(SRC_WS))
            setbflag(DST_WS);

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

        exit;
    }

When a call is placed, based on the two flags (STR_WS and DST_WS) we can determine what our caller and callee can "speak" (either RTP or DTLS-SRTP) and instruct the rtpengine daemon how to handle the call. We can do that by tuning the parameters passed to the rtpengine_offer() function.

    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)");

The rtpengine_answer() function logic should look like this:

    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)");

Now, all we have to do is to close the RTP/SRTP session when the call is ended. To do that, we use the rtpengine_delete() function:

    if (is_method("BYE|CANCEL")) {                                                                                                                      
        rtpengine_delete();

Having done all these settings should provide a full setup for interconnecting SIP clients over both UDP, TCP, etc. protocols, as well as browser based SIP clients over WebSocket.

조회 수 :
317
등록일 :
2017.09.06
08:19:51 (*.160.88.18)
엮인글 :
http://webs.co.kr/index.php?document_srl=3311817&act=trackback&key=701
게시글 주소 :
http://webs.co.kr/index.php?document_srl=3311817
List of Articles
번호 제목 글쓴이 날짜 조회 수
140 openssl 을 이용한 인증서 생성 절차를 정리한다. 개인키 CSR SSL 인증서 파일 생성 admin 2017-09-14 165
139 Documentation -> Tutorials -> TLS opensips.cfg admin 2017-09-14 170
138 Using TLS in OpenSIPS v2.2.x admin 2017-09-14 166
137 opensips tls cfg admin 2017-09-14 158
136 How to setup a Jabber / XMPP server on Debian 8 (jessie) using ejabberd admin 2017-09-13 212
135 SIP to XMPP Gateway + SIP Presence Server opensips admin 2017-09-13 205
134 OpenSIPS command line tricks admin 2017-09-13 209
133 Fail2Ban Freeswitch How to secure admin 2017-09-12 210
132 opensips.cfg. sample admin 2017-09-12 241
131 Advanced SIP scenarios with Event-based-Routing admin 2017-09-11 261
130 PUSH SERVER 푸시서버 안드로이드 애플 admin 2017-09-11 241
129 오픈소스 (사내)메신저 서버 구축, 오픈 파이어(openfire) 설치방법과 세팅(리눅스 기준) admin 2017-09-09 276
» rtpengine config basic and opensips configuration and command admin 2017-09-06 317
127 WebSocket Transport using OpenSIPS configuration 웹 소켓 컨피그레이션 기본 admin 2017-09-06 309
126 OpenSIPS basic configuration script 기본 컨피그 admin 2017-09-05 319
125 rtpengine install and config admin 2017-09-05 321
124 Installing RTPEngine on Ubuntu 14.04 admin 2017-09-05 328
123 compile only the textops module make modules=modules/textops modules admin 2017-09-05 341
122 opensips command /sbin/opensipsctl detail admin 2017-09-04 343
121 2017 08 31 opensips 2.32 install debian8.8 module install compile err modules admin 2017-09-04 341
120 Build-Depends debian 8.8 opensips 2.3 admin 2017-09-04 339
119 What is new in 2.3.0 opensips admin 2017-09-04 365
118 ubuntu 安装配置opensips,rtpproxy,mediaproxy admin 2017-09-04 353
117 How to install Mediaproxy 2.5.2 on CentOS 6 64 bit admin 2017-09-04 355
116 Using TLS in OpenSIPS v2.2.x configuration admin 2017-09-04 352
115 How to 2.3 download , OpenSIPS new apt repository. DEBs for Debian / Ubuntu admin 2017-09-02 358
114 You can install CDRTool in the following ways: admin 2017-09-01 370
113 How to Install OpenSIPS 2.1.2 Server on Ubuntu 15.04 admin 2017-09-01 366
112 Opensips 2.32 download admin 2017-09-01 371
111 OpenSIPS 2.3 install admin 2017-09-01 360
110 JsSIP: The JavaScript SIP Library admin 2017-09-01 363
109 WebSocket Transport using OpenSIPS admin 2017-09-01 380
108 A2Billing and OpenSIPS – Part 1 admin 2017-08-29 406
107 A2Billing and OpenSIPS – Part 2 admin 2017-08-29 405
106 A2Billing and OpenSIPS – Part 3 admin 2017-08-29 398
105 OpenSIPS 2.3 philosophy admin 2017-08-17 462
104 The timeline for OpenSIPS 2.3 is admin 2017-08-17 470
103 OpenSIPS Control Panel and Homer integration admin 2017-08-17 459
102 Opensips sip capture re designed admin 2017-07-16 599
101 WebRTC with OpenSIPS WebSocket is a protocol provides full-duplex admin 2015-04-04 5343
100 WebSocket Support in OpenSIPS 2.1 admin 2015-04-04 5981
99 OpenSIPS 2.1 (rc) is available, download now! admin 2015-03-22 5608
98 Service Provision Using Asterisk & OpenSIPS - AstriCon 2014 admin 2015-02-25 7142
97 SIP Signaling-Messages OpenSIPS Running On Multicore Server file admin 2014-11-02 14850
96 opensips.cfg for Asterisk admin 2014-10-20 16946
95 A2Billing and OpenSIPS config admin 2014-10-20 16319
94 Jitsi Videobridge meets WebRTC admin 2014-10-18 16298
93 A Survey of Open Source Products for Building a SIP Communication Platform admin 2014-10-18 16011
92 Script Function , Module Index v1.11 함수 모듈 opensips admin 2014-10-14 16172
91 Opensips TM module enables stateful processing of SIP transactions admin 2014-10-04 13927
90 kamailio.cfg configuration Example admin 2014-10-04 15993
89 opensips NAT Traversal Module admin 2014-10-02 15452
88 UAC Registrant Module admin 2014-09-28 16875
87 MediaProxy 2.3.x & OpenSIPS 1.5.x Integration admin 2014-08-24 16002
86 RTPPROXY Admin Guide admin 2014-08-24 16371
85 CANCEL MESSAGE not handled correctly admin 2014-08-23 16207
84 [Sipdroid] SIP data collection study tour admin 2014-08-23 16801
83 [OpenSIPS-Users] Opensips 1.10 NAT radius aaa admin 2014-08-23 16799
82 OpenSIPS Consultancy Pricing module install Server 판매 또는 설치및 컨설팅 가이드 admin 2014-08-23 16624
81 ICE: The ultimate way of beating NAT in SIP admin 2014-08-23 16470
80 Many OPENSIPS Configuration Examples This will Help you admin 2014-08-23 16160
79 Real-time Charging System for Telecom & ISP environments admin 2014-08-23 16758
78 OPENSIPS EBOOK admin 2014-08-21 16708
77 Opensips Documentation Function admin 2014-08-21 16938
76 Presence Tutorial OpenXCAP setup admin 2014-08-18 15965
75 Opensips Modules Documentation admin 2014-08-18 16809
74 A lightweight RPC library based on XML and HTTP admin 2014-08-18 16337
73 opensips Nat script with RTPPROXY - English Good perfect admin 2014-08-15 14387
72 OpenSIPS Control Panel (OCP) Installation Guide Good admin 2014-08-13 14338
71 Installation and configuration process record opensips opensips-cp admin 2014-08-13 22451
70 OpenSIPS as Homer Capture server admin 2014-08-13 14125
69 OpenSIPS , default script , Types of Routs , Routing in SIP, Video lecture admin 2014-08-13 16082
68 Configuracion de Kamailio 3.3 con NAT Traversal y XCAP. admin 2014-08-12 16468
67 Under RHEL6.5 install OpenSIPS 1.11.1 tls admin 2014-08-12 15370
66 OpenSIPS/OpenSER-a versatile SIP Server cfg admin 2014-08-11 16861
65 Kamailio Nat Traversal using RTPProxy admin 2014-08-11 16489
64 MediaProxy wiki page install configuration admin 2014-08-11 16458
63 오픈소스 (사내)메신저 서버 구축, 오픈 파이어(openfire) 설치방법과 세팅 admin 2014-08-11 24594
62 MediaProxy Installation Guide admin 2014-08-10 16034
61 RTPProxy 1.2.x Installation & Integration with OpenSIPS 1.5x admin 2014-08-10 17201
60 Opensips Installation, How to. Good guide wiki page admin 2014-08-10 14107
59 OpenSIPS Installation Notes admin 2014-08-09 13257
58 Installation and configuration process record opensips 1.9.1 admin 2014-08-09 15652
57 opensips 1.11.2 install Good Giide admin 2014-08-09 15584
56 fusionPBX install debian wheezy admin 2014-08-09 16324
55 opensips 1.11.2 install guide good 인스톨 가이드 admin 2014-08-09 15684
54 SigIMS IMS Platform admin 2014-05-24 17097
53 2013 2012년 분야별 최고의 오픈소스 소프트웨어 124선 admin 2014-04-05 16162
52 Video conference server OpenMCU-ru - Introduction admin 2014-04-01 19133
51 SIPSorcery admin 2014-03-18 17275
50 Ekiga (formely known as GnomeMeeting) is an open source SoftPhone admin 2014-03-12 17664
49 telepresence: Open Source SIP Telepresence/MCU admin 2014-03-12 19311
48 SIP PBX - OpenSIPS and Asterisk configuration admin 2014-03-12 15215
47 Conference Support in Kamailio (OpenSER) admin 2014-03-12 17713
46 OpenSIPS configuration for 2 or more FreeSWITCH installs admin 2014-03-12 15198
45 The Impact of TLS on SIP Server Performance file admin 2014-03-12 17556
44 book-opensips-101 / content / 3.2. SIP TLS Secure Calling.mediawiki admin 2014-03-12 16377
43 Where to check OpenSIPS does not start? admin 2014-03-09 17000
42 opensips-1.10.0_src.tar.gz experimental source code documentation admin 2014-03-09 17912
41 Kamailo OpenSIPs installation on Debian admin 2014-03-09 15983
40 Using the openSIPS Registrant Module admin 2014-03-09 17829
39 RTPproxy Frequentry Asked Questions (FAQ) ¶ admin 2014-03-07 15879
38 Building Telephony Systems with OpenSIPS 1.6 RTPProxy + OpenSIPS 1.7 admin 2014-03-07 17394
37 Installing RTPproxy Start RTPproxy in Bridged mode very good admin 2014-03-07 20895
36 OpenSIPS Control Panel (OCP) Installation Guide admin 2014-03-06 15755
35 OpenSIPS Control Panel install guide admin 2014-03-06 16555
34 rtpproxy Module admin 2014-03-06 17406
33 MediaProxy Installation Guide admin 2014-03-06 18135
32 How to install OpenSIPS on CentOS debian module add xcap admin 2014-03-06 17968
31 Problem with presence_xml module Opensips 1.9 admin 2014-03-06 17531
30 Building Telephony Systems with OpenSIPS 1.6 books file admin 2014-03-06 18323
29 Multimedia Service Platform admin 2014-03-06 16837
28 How to install OpenSIPS on CentOS Debian etc admin 2014-03-05 17847
27 Opensips Installation, How to. admin 2014-03-05 14213
26 100% CPU usage opensips admin 2014-03-05 17054
25 A2Billing and OpenSIPS admin 2014-03-04 17962
24 Opensips_1.9 install guide this is great I like this admin 2014-03-04 22038
23 Opensips install debian admin 2014-03-03 18103
22 Open Source VOIP applications, both clients and servers. admin 2013-11-20 18398
21 OfficeSIP Server is freeware VoIP, SIP server for Windows admin 2013-09-11 19053
20 My new toy: Bluebox-ng admin 2013-04-06 31997
19 Flooding Asterisk, Freeswitch and Kamailio with Metasploit admin 2013-04-06 27713
18 Asterisk Installation Asterisk Realtime configuration admin 2013-04-06 21818
17 The SIP Router Project admin 2013-04-06 20892
16 Kamailio :: A Quick Introduction admin 2013-04-06 17923
15 Welcome to the Smartvox Knowledgebase admin 2013-04-06 18665
14 Kamailio 3.3.x and Asterisk 10.7.0 Realtime Integration using Asterisk Database admin 2013-04-06 21501
13 OpenSIPS vs Asterisk admin 2013-04-06 26783
12 OpenSER_from_an_asterisk_POV file admin 2013-04-06 18635
11 Using SIP Devices behind NAT OPensip Asterisk IPPhone SIP Telephony file admin 2013-03-31 34249
10 rfc5766-turn-server admin 2013-03-21 20462
9 OpenSIPS Kick Start‎: VIDEO admin 2013-02-20 17870
8 OPENSIP Training VIDEO admin 2013-02-20 17790
7 What is new in 1.8.0 opensip admin 2012-05-21 38916
6 Asterisk v1.4x built on FreeBSD v7.1 UNIX admin 2012-01-06 28959
5 SIP 트래픽 생성 테스트 툴 admin 2011-12-23 37965
4 사설 망 환경에서 SIP 의 NAT Traversal 문제 admin 2011-12-23 32676
3 the OpenSIPS Project OpenSIP admin 2011-12-14 17844
2 OpenH323 Gatekeeper - The GNU Gatekeeper admin 2011-12-14 22608
1 The FreeRADIUS Project admin 2011-12-14 19086