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


http://kb.smartvox.co.uk/opensips/opensips-asterisk/


OpenSIPS and Asterisk are both open source projects and both are used for Voice over IP. However, they perform quite different roles, have different capabilities and different strengths and weaknesses. This article reviews how they are so different and considers what role each product can play in the infrastructure of an Internet Telephony Service Provider solution.

Fundamental differences

A succinct, but slightly technical, distinction between OpenSIPS and Asterisk is that OpenSIPS is essentially a SIP Proxy Server while Asterisk is essentially a Media Server (for a detailed explanation of these terms click here). This means that OpenSIPS is not normally the final end-point for a voice call; instead it relays the call control signals from one server to another. On the other hand, Asterisk does terminate calls and, even if it appears to relay a call onward to another destination, it does so be creating a new call and linking the audio streams to make the two calls appear as one – this behaviour is referred to as a “Back-to-Back User Agent” or B2BUA for short.

There are several other important differences including:

  • OpenSIPS has no inbuilt media capabilities – it cannot record or play speech or music
  • Asterisk has media capabilities which means it can be used as an IVR (with menus of options for callers to select using their keypad) or as a voicemail server etc.
  • Asterisk has many built-in features that allow it to act as an IP-PBX or business telephone system. It can also act as a conference server. OpenSIPS does not have these capabilities.
  • Asterisk can act as a gateway between different telephony technologies (including ISDN) whereas OpenSIPS is only a SIP Server.

So, you may think OpenSIPS sounds like a poor relation of Asterisk with fewer capabilities, but in fact these products are just not designed to do the same job. Where OpenSIPS scores is in its robustness, flexibility, adherence to SIP standards and, above all, its speed and call capacity which are in a completely different league to those of Asterisk.

Links to more detailed point-by-point comparison tables can be found under the heading of “further reading” at the end of this article.

When to choose Asterisk

Asterisk can interconnect with legacy telephony technologies such as analogue (FXS ports where handsets are to be connected and FXO where a line from the Telco needs to be connected) and ISDN. Combined with its built-in PBX features, this makes it very suitable for customer premises as a replacement for, or add-on to, an existing PBX or business telephone system. Its voicemail, IVR and conferencing capabilities make it a popular choice for hosted service providers and its ability to bridge different telephone technologies means it can also be used as a PSTN gateway, for example where a service provider wants a resilient infrastructure with multiple disparate paths connecting their services to the public telephone network.

For many applications, especially as corporate end-user equipment or CPE, Asterisk is very competent. In these situations, I would generally recommend using a distribution that includes a web GUI interface (FreePBX, PBX-in-a-flash, Elastix or similar). However, Asterisk has some weaknesses and limitations that become more apparent as your solution is scaled up. For a telephony service provider, the call capacity of a single Asterisk server – typically up to 200 simultaneous calls – is likely to be too low. In a complex network environment, its slightly simplistic solutions for near-end and far-end NAT traversal may not work for some situations or for some makes of IP phone or ATA’s. It is a popular target for hackers, who know all its loopholes and weaknesses. From experience, I also have doubts about its ability to stand up to denial of service attacks. It does not support multiple SIP registrations to the same account.

When to choose OpenSIPS

OpenSIPS is frequently used by Internet telephony service providers (ITSP’s) as a “front door” – a connection point for a wide range of SIP devices and SIP trunks. In the right hands, it can be configured to cope with a range of different network architectures and can identify and fix non-standard SIP implementations as well as being able to fix the SIP headers for remote devices operating behind NAT. It can be used to load balance incoming calls to a group of media servers- possibly Asterisk servers. For outbound traffic, it can select different carriers using least cost routing (LCR) combined with failover in case the preferred route is unavailable or already at maximum capacity. Unlike Asterisk, it correctly handles multiple registrations to the same account.

The flexibility of OpenSIPS derives from its use of a configuration file containing a control script written like a computer program – the language is similar to C. This allows the application to inspect the contents of each SIP request down to a very low level, looking at headers, source address, contact address, user-agent, call ID, etc. The SIP requests can be modified, forwarded, forked, dropped or sent a response. In this way it is easy for OpenSIPS to distinguish between normal traffic and illegal requests or requests designed to identify open ports on SIP servers (e.g. friendly-scanner). This ability makes OpenSIPS even more suitable as a “front door” for access to to your other SIP-based servers – it has the ability to recognise and drop the most common port-scanning requests, the capacity and robustness to withstand DoS attacks and the flexibility to let you add rules for special cases and exceptions. What I am describing is, almost, a SIP firewall. Viewed in this light, OpenSIPS is a direct alternative to the so-called Session Border Controllers (SBC’s) that are increasingly being promoted as a “must-have” component in corporate VoIP solutions.

When combined with Mediaproxy from AG-Projects, OpenSIPS is able to solve problems of far-end NAT traversal in a more refined manner than Asterisk. The structure and content of external data referenced by OpenSIPS at run time (e.g. tables in a MySQL database) is better defined than the equivalent within Asterisk. OpenSIPS adheres more closely to the RFC’s that define how SIP should behave and it can support SIP over TCP or UDP as well as secure SIP over TLS. However, on the minus side, OpenSIPS cannot do codec translation (transcoding) or use telephony cards to interface to ISDN, analogue etc, it does not support IAX or H323, and it has no media capabilities for recording, playing voice files, mixing, conferencing, text-to-speech or speech recognition.

Teaching OpenSIPS new tricks

As a product, OpenSIPS does not stand still. Version 1.7 was released in the summer of 2011 and further releases are promised soon. By virtue of the Dialog module, it is now fully state aware which has allowed many improvements including better Call Data Records (CDR’s) and the ability to limit the number of simultaneous calls allowed to a particular end point. The developers have also added a B2BUA module. In part the justification for this was to allow improved topology hiding – the process whereby the addresses of some of your telephony servers can be hidden from the external users. This may be of benefit where OpenSIPS is required to act as a kind of SIP firewall or SBC, but it seems somewhat out of character to me, especially as one of the original oft-quoted differences between Asterisk and OpenSIPS was that Asterisk, unlike OpenSIPS, is a B2BUA.

Working together

In my experience, there are great benefits to be gained from using a combination of OpenSIPS and Asterisk. They complement each other well, especially for the design of a secure, scalable, high volume hosted telephony solution. OpenSIPS can be used as the main portal and can load balance incoming SIP requests to multiple Asterisk boxes. In this role, OpenSIPS is also able to protect the Asterisk servers from the majority of port scanning and password guessing intrusions. Another – or possibly the same – OpenSIPS server can route outbound SIP requests to carriers using LCR.  The outbound routing can be augmented with automatic probing of routes to monitor for faults so that traffic can be redirected to another destination when a fault is detected.

Our experience has shown that OpenSIPS combined with Asterisk presents some new and interesting challenges. For example, you have to decide if registrations are to be made to OpenSIPS only, Asterisk only or to both. Furthermore, authentication challenges and their responses, when initiated by Asterisk, must be allowed to pass through OpenSIPS in a transparent manner. Other issues we have solved at Smartvox include proper handling of keep-alive packets, SIP session timers, routing and SDP address fixing for servers behind NAT or in a DMZ, address substitutions for mixed carrier connections (some carriers using private peering and others on the Internet), fixing of non-symmetric SIP ports for Cisco handsets, Asterisk SIP timer issues, activation of message waiting lamps and presence. We are always happy to gain new clients and take on new challenges, so please get in touchto discuss your project.

Further reading

An excellent point-by-point comparison between OpenSIPS and Asterisk is published on the web site of “VOIP Today”. Click here to open their article.

Another point-by-point comparison was written in 2008 in an article that I believe should be attributed to Flavio Goncalves (although the same text has since been copied and re-published many times by others pretending it is their own work). Here is a link to the article.


조회 수 :
72780
등록일 :
2013.04.06
22:25:39 (*.160.42.88)
엮인글 :
http://webs.co.kr/index.php?document_srl=19753&act=trackback&key=64b
게시글 주소 :
http://webs.co.kr/index.php?document_srl=19753
List of Articles
번호 제목 글쓴이 날짜 조회 수
162 Opensips Gateway between SIP and SMPP messages admin 2019-02-19 281
161 smpp sms opensips admin 2019-02-19 264
160 Busy Lamp Field (BLF) feature on Opensips 2.4.0 with Zoiper configuration admin 2018-05-29 2006
159 Documentation -> Tutorials -> WebSocket Transport using OpenSIPS admin 2018-05-17 1888
158 List of SIP response codes admin 2017-12-20 3545
157 opensips/modules/event_routing/ Push Notification Call pickup admin 2017-12-20 3104
156 opensips push notification How to detail file admin 2017-12-20 3024
155 OpenSIPS routing logic admin 2017-12-12 3071
154 OpenSIPS example configuration admin 2017-12-12 3064
153 opensips log output admin 2017-12-11 3059
152 opensips complete configuration example admin 2017-12-10 3170
151 Opensips1.6 ebook detail configuration and SIP signal and NAT etc file admin 2017-12-10 3141
150 dictionary.opensips radius admin 2017-12-09 4093
149 what is record_route() in opensips ? admin 2017-12-09 4023
148 what is loose_route() in opensips ? file admin 2017-12-09 4178
147 in opensips what is lookup(domain [, flags [, aor]]) admin 2017-12-09 4046
146 in opensips db_does_uri_exist() what is admin 2017-12-09 3887
145 in opensips what is has_totag() admin 2017-12-09 4056
144 opensips exec module admin 2017-12-08 4229
143 opensips push notification How to admin 2017-12-07 4031
142 OpenSIPS Module Interface admin 2017-12-07 4139
141 opensips configuration config explain easy basic 오픈쉽스 컨피그레이션 기본 설명 file admin 2017-12-07 4212
140 openssl 을 이용한 인증서 생성 절차를 정리한다. 개인키 CSR SSL 인증서 파일 생성 admin 2017-09-14 5306
139 Documentation -> Tutorials -> TLS opensips.cfg admin 2017-09-14 5070
138 Using TLS in OpenSIPS v2.2.x admin 2017-09-14 5064
137 opensips tls cfg admin 2017-09-14 5151
136 How to setup a Jabber / XMPP server on Debian 8 (jessie) using ejabberd admin 2017-09-13 5700
135 SIP to XMPP Gateway + SIP Presence Server opensips admin 2017-09-13 5024
134 OpenSIPS command line tricks admin 2017-09-13 4993
133 Fail2Ban Freeswitch How to secure admin 2017-09-12 5275
132 opensips.cfg. sample admin 2017-09-12 4958
131 Advanced SIP scenarios with Event-based-Routing admin 2017-09-11 5130
130 PUSH SERVER 푸시서버 안드로이드 애플 admin 2017-09-11 5589
129 오픈소스 (사내)메신저 서버 구축, 오픈 파이어(openfire) 설치방법과 세팅(리눅스 기준) admin 2017-09-09 13890
128 rtpengine config basic and opensips configuration and command admin 2017-09-06 5302
127 WebSocket Transport using OpenSIPS configuration 웹 소켓 컨피그레이션 기본 admin 2017-09-06 5122
126 OpenSIPS basic configuration script 기본 컨피그 admin 2017-09-05 5223
125 rtpengine install and config admin 2017-09-05 5199
124 Installing RTPEngine on Ubuntu 14.04 admin 2017-09-05 5308
123 compile only the textops module make modules=modules/textops modules admin 2017-09-05 5177
122 opensips command /sbin/opensipsctl detail admin 2017-09-04 5276
121 2017 08 31 opensips 2.32 install debian8.8 module install compile err modules admin 2017-09-04 5203
120 Build-Depends debian 8.8 opensips 2.3 admin 2017-09-04 5097
119 What is new in 2.3.0 opensips admin 2017-09-04 5929
118 ubuntu 安装配置opensips,rtpproxy,mediaproxy admin 2017-09-04 5464
117 How to install Mediaproxy 2.5.2 on CentOS 6 64 bit admin 2017-09-04 5690
116 Using TLS in OpenSIPS v2.2.x configuration admin 2017-09-04 5357
115 How to 2.3 download , OpenSIPS new apt repository. DEBs for Debian / Ubuntu admin 2017-09-02 5317
114 You can install CDRTool in the following ways: admin 2017-09-01 5623
113 How to Install OpenSIPS 2.1.2 Server on Ubuntu 15.04 admin 2017-09-01 5527
112 Opensips 2.32 download admin 2017-09-01 5306
111 OpenSIPS 2.3 install admin 2017-09-01 5628
110 JsSIP: The JavaScript SIP Library admin 2017-09-01 5575
109 WebSocket Transport using OpenSIPS admin 2017-09-01 5665
108 A2Billing and OpenSIPS – Part 1 admin 2017-08-29 5367
107 A2Billing and OpenSIPS – Part 2 admin 2017-08-29 5274
106 A2Billing and OpenSIPS – Part 3 admin 2017-08-29 5488
105 OpenSIPS 2.3 philosophy admin 2017-08-17 6024
104 The timeline for OpenSIPS 2.3 is admin 2017-08-17 6161
103 OpenSIPS Control Panel and Homer integration admin 2017-08-17 6202
102 Opensips sip capture re designed admin 2017-07-16 5650
101 WebRTC with OpenSIPS WebSocket is a protocol provides full-duplex admin 2015-04-04 10673
100 WebSocket Support in OpenSIPS 2.1 admin 2015-04-04 11994
99 OpenSIPS 2.1 (rc) is available, download now! admin 2015-03-22 10558
98 Service Provision Using Asterisk & OpenSIPS - AstriCon 2014 admin 2015-02-25 12367
97 SIP Signaling-Messages OpenSIPS Running On Multicore Server file admin 2014-11-02 19895
96 opensips.cfg for Asterisk admin 2014-10-20 22138
95 A2Billing and OpenSIPS config admin 2014-10-20 21457
94 Jitsi Videobridge meets WebRTC admin 2014-10-18 23116
93 A Survey of Open Source Products for Building a SIP Communication Platform admin 2014-10-18 21078
92 Script Function , Module Index v1.11 함수 모듈 opensips admin 2014-10-14 21289
91 Opensips TM module enables stateful processing of SIP transactions admin 2014-10-04 18975
90 kamailio.cfg configuration Example admin 2014-10-04 21247
89 opensips NAT Traversal Module admin 2014-10-02 20535
88 UAC Registrant Module admin 2014-09-28 22301
87 MediaProxy 2.3.x & OpenSIPS 1.5.x Integration admin 2014-08-24 21472
86 RTPPROXY Admin Guide admin 2014-08-24 21823
85 CANCEL MESSAGE not handled correctly admin 2014-08-23 21631
84 [Sipdroid] SIP data collection study tour admin 2014-08-23 22042
83 [OpenSIPS-Users] Opensips 1.10 NAT radius aaa admin 2014-08-23 22036
82 OpenSIPS Consultancy Pricing module install Server 판매 또는 설치및 컨설팅 가이드 admin 2014-08-23 21938
81 ICE: The ultimate way of beating NAT in SIP admin 2014-08-23 21581
80 Many OPENSIPS Configuration Examples This will Help you admin 2014-08-23 21238
79 Real-time Charging System for Telecom & ISP environments admin 2014-08-23 21991
78 OPENSIPS EBOOK admin 2014-08-21 22138
77 Opensips Documentation Function admin 2014-08-21 21829
76 Presence Tutorial OpenXCAP setup admin 2014-08-18 21426
75 Opensips Modules Documentation admin 2014-08-18 22101
74 A lightweight RPC library based on XML and HTTP admin 2014-08-18 21274
73 opensips Nat script with RTPPROXY - English Good perfect admin 2014-08-15 20078
72 OpenSIPS Control Panel (OCP) Installation Guide Good admin 2014-08-13 20285
71 Installation and configuration process record opensips opensips-cp admin 2014-08-13 46507
70 OpenSIPS as Homer Capture server admin 2014-08-13 19188
69 OpenSIPS , default script , Types of Routs , Routing in SIP, Video lecture admin 2014-08-13 21364
68 Configuracion de Kamailio 3.3 con NAT Traversal y XCAP. admin 2014-08-12 21845
67 Under RHEL6.5 install OpenSIPS 1.11.1 tls admin 2014-08-12 21135
66 OpenSIPS/OpenSER-a versatile SIP Server cfg admin 2014-08-11 22007
65 Kamailio Nat Traversal using RTPProxy admin 2014-08-11 21563
64 MediaProxy wiki page install configuration admin 2014-08-11 21620
63 오픈소스 (사내)메신저 서버 구축, 오픈 파이어(openfire) 설치방법과 세팅 admin 2014-08-11 40413
62 MediaProxy Installation Guide admin 2014-08-10 21134
61 RTPProxy 1.2.x Installation & Integration with OpenSIPS 1.5x admin 2014-08-10 22371
60 Opensips Installation, How to. Good guide wiki page admin 2014-08-10 19384
59 OpenSIPS Installation Notes admin 2014-08-09 18898
58 Installation and configuration process record opensips 1.9.1 admin 2014-08-09 32450
57 opensips 1.11.2 install Good Giide admin 2014-08-09 22455
56 fusionPBX install debian wheezy admin 2014-08-09 21328
55 opensips 1.11.2 install guide good 인스톨 가이드 admin 2014-08-09 21626
54 SigIMS IMS Platform admin 2014-05-24 21878
53 2013 2012년 분야별 최고의 오픈소스 소프트웨어 124선 admin 2014-04-05 26333
52 Video conference server OpenMCU-ru - Introduction admin 2014-04-01 24562
51 SIPSorcery admin 2014-03-18 22306
50 Ekiga (formely known as GnomeMeeting) is an open source SoftPhone admin 2014-03-12 22727
49 telepresence: Open Source SIP Telepresence/MCU admin 2014-03-12 47065
48 SIP PBX - OpenSIPS and Asterisk configuration admin 2014-03-12 35547
47 Conference Support in Kamailio (OpenSER) admin 2014-03-12 29968
46 OpenSIPS configuration for 2 or more FreeSWITCH installs admin 2014-03-12 20902
45 The Impact of TLS on SIP Server Performance file admin 2014-03-12 22342
44 book-opensips-101 / content / 3.2. SIP TLS Secure Calling.mediawiki admin 2014-03-12 21579
43 Where to check OpenSIPS does not start? admin 2014-03-09 21673
42 opensips-1.10.0_src.tar.gz experimental source code documentation admin 2014-03-09 22713
41 Kamailo OpenSIPs installation on Debian admin 2014-03-09 28416
40 Using the openSIPS Registrant Module admin 2014-03-09 23177
39 RTPproxy Frequentry Asked Questions (FAQ) ¶ admin 2014-03-07 21136
38 Building Telephony Systems with OpenSIPS 1.6 RTPProxy + OpenSIPS 1.7 admin 2014-03-07 22220
37 Installing RTPproxy Start RTPproxy in Bridged mode very good admin 2014-03-07 35997
36 OpenSIPS Control Panel (OCP) Installation Guide admin 2014-03-06 20885
35 OpenSIPS Control Panel install guide admin 2014-03-06 22121
34 rtpproxy Module admin 2014-03-06 22007
33 MediaProxy Installation Guide admin 2014-03-06 30366
32 How to install OpenSIPS on CentOS debian module add xcap admin 2014-03-06 22891
31 Problem with presence_xml module Opensips 1.9 admin 2014-03-06 22391
30 Building Telephony Systems with OpenSIPS 1.6 books file admin 2014-03-06 23398
29 Multimedia Service Platform admin 2014-03-06 21709
28 How to install OpenSIPS on CentOS Debian etc admin 2014-03-05 22489
27 Opensips Installation, How to. admin 2014-03-05 19182
26 100% CPU usage opensips admin 2014-03-05 21897
25 A2Billing and OpenSIPS admin 2014-03-04 23715
24 Opensips_1.9 install guide this is great I like this admin 2014-03-04 29215
23 Opensips install debian admin 2014-03-03 23025
22 Open Source VOIP applications, both clients and servers. admin 2013-11-20 23413
21 OfficeSIP Server is freeware VoIP, SIP server for Windows admin 2013-09-11 24619
20 My new toy: Bluebox-ng admin 2013-04-06 39010
19 Flooding Asterisk, Freeswitch and Kamailio with Metasploit admin 2013-04-06 41498
18 Asterisk Installation Asterisk Realtime configuration admin 2013-04-06 27505
17 The SIP Router Project admin 2013-04-06 26455
16 Kamailio :: A Quick Introduction admin 2013-04-06 23978
15 Welcome to the Smartvox Knowledgebase admin 2013-04-06 24288
14 Kamailio 3.3.x and Asterisk 10.7.0 Realtime Integration using Asterisk Database admin 2013-04-06 29200
» OpenSIPS vs Asterisk admin 2013-04-06 72780