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


Gateway between SIP and SMPP messages

phone

While the SIP protocol is one of the most popular protocols used for voice calls, the SMPP (Short Message Peer-to-Peer) is one of the most widely used protocols for sending text messages. Having both of them offered by your service enhances your platform with more compatibility and flexibility.

In order for your customers to have an unified experience, one has to ensure that messages coming from one side are visible and accessible by the other side. For example, someone might send an SMS that comes to your platform over SMPP and it needs to be delivered to a SIP endpoint. Since the two protocols are incompatible (see Compatibility below), there needs to be a tool that does bridging between the two protocols.

Luckily, OpenSIPS 3.0 now features a SIP to SMPP gateway using the new proto_smpp module! This module acts as an ESMEs (External Short Messaging Entity) that is able to connect to a SMSCs (Short Message Service Center) and exchange text messages with it.

Compatibility

Messages in SMPP are binary encoded PDUs (protocol data units) that are carried between the entities involved over TCP. Since it’s using a stream oriented transport protocol, the SMPP protocol has a plethora of commands that are used to keep a SMPP session active and exchange data between endpoints: when starting a SMPP session, an ESME needs to send one of four bind commands (bind_receiver, bind_transmitter, bind_transceiver, or outbind) to the SMSC, depending on its role (receiver, transmitter, or both). Each command has to be confirmed (for example using a bind_receiver_resp acknowledgement for a bind_receiver command). After that, pinging (enquire_link) has to be sent periodically to keep the connection open. Finally, when a text message is sent, a different command command is used, depending on the state of the component.

On the other hand, in SIP things are simpler – messages are constructed using plain text SIP MESSAGE packets (RFC 3428). These messages can be exchanged between SIP endpoints and proxies using any transport protocol supported by SIP (UDP, TCP, TLS, SCTP, WebSocket), the most common one being UDP. Each SIP message sent needs to be confirmed by the receiver using a 200 OK. And that’s it.

Therefore it is clear that the two protocols are not compatible from a behavioral point of view. However, at the end of the day, both protocols are used to transport (mainly) text messages. So as long as we have the actual message payload and meta-data, all we have to do is to building the message according to the transport protocol we need to use.

Configuration

In order to use SMPP connections in your OpenSIPS install, you have to load the proto_smpp module and define a listener that will be used for communication:

listen = smpp:127.0.0.1:2775
...
load_module "proto_smpp.so"

As noted in previous section, for an SMSC to be able to deliver messages to an ESME, all SMPP connections need to be created beforehand. This is done automatically by OpenSIPS at startup. All connections that have to be initiated are described in a database, along with all the parameters they need. So the next thing that has to be configured in the script is a connection to a database:

mysql://opensips:opensipsrw@localhost/opensips

Below you can find an example of a MySQL record that describes an SMPP connection used to connect to an SMSC as a transceiver (session_type=1):

        name: SMPP_test
          ip: 127.0.0.1
        port: 2777
   system_id: smpp
    password: test
 system_type: 
     src_ton: 2
     src_npi: 1
     dst_ton: 2
     dst_npi: 1
session_type: 1
  • name: represents an arbitrary, unique name that will be used to reference this SMPP connection from the script
  • ip and port: the TCP information needed to connect to the SMSC
  • system_id (also known as the user name) and password are used for authentication
  • system_type is a field required by some SMPP providers, and it is usually used to identify the types of services that connection is allowed to use
  • TON (Type of Number) for source (src_ton) and destination (dst_ton) indicate the format of the numbers used for sender and receiver. Some common values are:
    • 0 – Unknown
    • 1 – International
    • 2 – National
    • 3 – Network Specific
    • 4 – Subscriber Number
    • 5 – Alphanumeric
    • 6 – Abbreviated
  • NPI (Number Plan Indicator) for source (src_npi) and destination (dst_npi) represent the numbering scheme used for the clients. Common values are:
    • 0 – Unknown
    • 1 – ISDN/telephone numbering plan (E163/E164)
    • 3 – Data numbering plan (X.121)
    • 4 – Telex numbering plan (F.69)
    • 6 – Land Mobile (E.212)
    • 8 – National numbering plan
    • 9 – Private numbering plan
    • 10 – ERMES numbering plan (ETSI DE/PS 3 01-3)
    • 13 – Internet (IP)
    • 18 – WAP Client Id (to be defined by WAP Forum)
  • session_type is used to specify the type of connection, and has to have one of the following values:
    • 1 – Transceiver
    • 2 – Transmitter
    • 3 – Receiver
    • 4 – Outbind

Once the connection has started, OpenSIPS can use it to exchange messages with its peer. The module will handle in the background all the SMPP protocol’s specifics.

SIP to SMPP gateway

When getting a text message from a SIP entity, it will be handled just like any other SIP request, by running the script. And if according your routing logic that message needs to end up to a SMSC, all you have to do is to call the send_smpp_message() method, specifying which SMSC should be used:

if (is_method("MESSAGE") && isflagset(TO_SMPP_TEST))
    send_smpp_message("SMPP_test");

SMPP to SIP gateway

When a message comes from a SMSC to OpenSIPS over SMPP, things get a bit more complicated – since SMPP messages are binary encoded, they can’t be sent directly to the script. Instead, they are translated to a SIP MESSAGE, and automatically sent to a SIP server or proxy, indicated by the outbound_uri parameter. Usually, the URI points to the same OpenSIPS instance, but note that the message will be sent over SIP – this means that it will be able to enter the script for determining its next hop, or final destination.

listen = udp:127.0.0.1:5060
...
# send all SMPP messages on loopback to the same instance
modparam("proto_smpp", "outbound_uri", "sip:127.0.0.1:5060")
...
route {
    ...
    if (is_method("MESSAGE") && $Ri == "127.0.0.1") {
        # handle message received over SMPP
        ...
    }
    ...
}

And that’s it, you know have a two-way gateway beween SIP and SMPP.

Conclusions

With only a few lines of configuration and the new SMPP module you can easily enhance your OpenSIPS-based VoIP platform with new means of sending text messages. Just give it a try and let us know if you ran into any trouble while setting up this feature!

Special thanks go to Victor Ciurel for developing and testing the new SMPP module in OpenSIPS!

조회 수 :
79
등록일 :
2019.02.19
22:15:06 (*.214.125.21)
엮인글 :
http://webs.co.kr/index.php?document_srl=3318883&act=trackback&key=93e
게시글 주소 :
http://webs.co.kr/index.php?document_srl=3318883
List of Articles
번호 제목 글쓴이 조회 수 추천 수 날짜
» Opensips Gateway between SIP and SMPP messages admin 79   2019-02-19
Gateway between SIP and SMPP messages While the SIP protocol is one of the most popular protocols used for voice calls, the SMPP (Short Message Peer-to-Peer) is one of the most widely used protoc...  
161 smpp sms opensips admin 76   2019-02-19
 
160 Busy Lamp Field (BLF) feature on Opensips 2.4.0 with Zoiper configuration admin 1784   2018-05-29
 
159 Documentation -> Tutorials -> WebSocket Transport using OpenSIPS admin 1650   2018-05-17
 
158 List of SIP response codes admin 3306   2017-12-20
 
157 opensips/modules/event_routing/ Push Notification Call pickup admin 2868   2017-12-20
 
156 opensips push notification How to detail file admin 2770   2017-12-20
 
155 OpenSIPS routing logic admin 2844   2017-12-12
 
154 OpenSIPS example configuration admin 2827   2017-12-12
 
153 opensips log output admin 2831   2017-12-11
 
152 opensips complete configuration example admin 2920   2017-12-10
 
151 Opensips1.6 ebook detail configuration and SIP signal and NAT etc file admin 2920   2017-12-10
 
150 dictionary.opensips radius admin 3838   2017-12-09
 
149 what is record_route() in opensips ? admin 3763   2017-12-09
 
148 what is loose_route() in opensips ? file admin 3884   2017-12-09
 
147 in opensips what is lookup(domain [, flags [, aor]]) admin 3797   2017-12-09
 
146 in opensips db_does_uri_exist() what is admin 3642   2017-12-09
 
145 in opensips what is has_totag() admin 3794   2017-12-09
 
144 opensips exec module admin 3971   2017-12-08
 
143 opensips push notification How to admin 3743   2017-12-07
 
142 OpenSIPS Module Interface admin 3880   2017-12-07
 
141 opensips configuration config explain easy basic 오픈쉽스 컨피그레이션 기본 설명 file admin 3929   2017-12-07
 
140 openssl 을 이용한 인증서 생성 절차를 정리한다. 개인키 CSR SSL 인증서 파일 생성 admin 5009   2017-09-14
 
139 Documentation -> Tutorials -> TLS opensips.cfg admin 4786   2017-09-14
 
138 Using TLS in OpenSIPS v2.2.x admin 4768   2017-09-14
 
137 opensips tls cfg admin 4901   2017-09-14
 
136 How to setup a Jabber / XMPP server on Debian 8 (jessie) using ejabberd admin 5407   2017-09-13
 
135 SIP to XMPP Gateway + SIP Presence Server opensips admin 4746   2017-09-13
 
134 OpenSIPS command line tricks admin 4719   2017-09-13
 
133 Fail2Ban Freeswitch How to secure admin 5000   2017-09-12
 
132 opensips.cfg. sample admin 4712   2017-09-12
 
131 Advanced SIP scenarios with Event-based-Routing admin 4858   2017-09-11
 
130 PUSH SERVER 푸시서버 안드로이드 애플 admin 5219   2017-09-11
 
129 오픈소스 (사내)메신저 서버 구축, 오픈 파이어(openfire) 설치방법과 세팅(리눅스 기준) admin 12593   2017-09-09
 
128 rtpengine config basic and opensips configuration and command admin 5012   2017-09-06
 
127 WebSocket Transport using OpenSIPS configuration 웹 소켓 컨피그레이션 기본 admin 4828   2017-09-06
 
126 OpenSIPS basic configuration script 기본 컨피그 admin 4963   2017-09-05
 
125 rtpengine install and config admin 4911   2017-09-05
 
124 Installing RTPEngine on Ubuntu 14.04 admin 5004   2017-09-05
 
123 compile only the textops module make modules=modules/textops modules admin 4903   2017-09-05
 
122 opensips command /sbin/opensipsctl detail admin 4988   2017-09-04
 
121 2017 08 31 opensips 2.32 install debian8.8 module install compile err modules admin 4940   2017-09-04
 
120 Build-Depends debian 8.8 opensips 2.3 admin 4820   2017-09-04
 
119 What is new in 2.3.0 opensips admin 5586   2017-09-04
 
118 ubuntu 安装配置opensips,rtpproxy,mediaproxy admin 5189   2017-09-04
 
117 How to install Mediaproxy 2.5.2 on CentOS 6 64 bit admin 5393   2017-09-04
 
116 Using TLS in OpenSIPS v2.2.x configuration admin 5071   2017-09-04
 
115 How to 2.3 download , OpenSIPS new apt repository. DEBs for Debian / Ubuntu admin 5042   2017-09-02
 
114 You can install CDRTool in the following ways: admin 5255   2017-09-01
 
113 How to Install OpenSIPS 2.1.2 Server on Ubuntu 15.04 admin 5252   2017-09-01
 
112 Opensips 2.32 download admin 5020   2017-09-01
 
111 OpenSIPS 2.3 install admin 5306   2017-09-01
 
110 JsSIP: The JavaScript SIP Library admin 5293   2017-09-01
 
109 WebSocket Transport using OpenSIPS admin 5378   2017-09-01
 
108 A2Billing and OpenSIPS – Part 1 admin 5101   2017-08-29
 
107 A2Billing and OpenSIPS – Part 2 admin 4997   2017-08-29
 
106 A2Billing and OpenSIPS – Part 3 admin 5214   2017-08-29
 
105 OpenSIPS 2.3 philosophy admin 5707   2017-08-17
 
104 The timeline for OpenSIPS 2.3 is admin 5882   2017-08-17
 
103 OpenSIPS Control Panel and Homer integration admin 5821   2017-08-17
 
102 Opensips sip capture re designed admin 5378   2017-07-16
 
101 WebRTC with OpenSIPS WebSocket is a protocol provides full-duplex admin 10357   2015-04-04
 
100 WebSocket Support in OpenSIPS 2.1 admin 11248   2015-04-04
 
99 OpenSIPS 2.1 (rc) is available, download now! admin 10249   2015-03-22
 
98 Service Provision Using Asterisk & OpenSIPS - AstriCon 2014 admin 11962   2015-02-25
 
97 SIP Signaling-Messages OpenSIPS Running On Multicore Server file admin 19571   2014-11-02
 
96 opensips.cfg for Asterisk admin 21764   2014-10-20
 
95 A2Billing and OpenSIPS config admin 21096   2014-10-20
 
94 Jitsi Videobridge meets WebRTC admin 22283   2014-10-18
 
93 A Survey of Open Source Products for Building a SIP Communication Platform admin 20725   2014-10-18
 
92 Script Function , Module Index v1.11 함수 모듈 opensips admin 20954   2014-10-14
 
91 Opensips TM module enables stateful processing of SIP transactions admin 18635   2014-10-04
 
90 kamailio.cfg configuration Example admin 20867   2014-10-04
 
89 opensips NAT Traversal Module admin 20185   2014-10-02
 
88 UAC Registrant Module admin 21935   2014-09-28
 
87 MediaProxy 2.3.x & OpenSIPS 1.5.x Integration admin 21079   2014-08-24
 
86 RTPPROXY Admin Guide admin 21458   2014-08-24
 
85 CANCEL MESSAGE not handled correctly admin 21222   2014-08-23
 
84 [Sipdroid] SIP data collection study tour admin 21674   2014-08-23
 
83 [OpenSIPS-Users] Opensips 1.10 NAT radius aaa admin 21680   2014-08-23
 
82 OpenSIPS Consultancy Pricing module install Server 판매 또는 설치및 컨설팅 가이드 admin 21577   2014-08-23
 
81 ICE: The ultimate way of beating NAT in SIP admin 21235   2014-08-23
 
80 Many OPENSIPS Configuration Examples This will Help you admin 20918   2014-08-23
 
79 Real-time Charging System for Telecom & ISP environments admin 21645   2014-08-23
 
78 OPENSIPS EBOOK admin 21767   2014-08-21
 
77 Opensips Documentation Function admin 21535   2014-08-21
 
76 Presence Tutorial OpenXCAP setup admin 21018   2014-08-18
 
75 Opensips Modules Documentation admin 21743   2014-08-18
 
74 A lightweight RPC library based on XML and HTTP admin 20960   2014-08-18
 
73 opensips Nat script with RTPPROXY - English Good perfect admin 19579   2014-08-15
 
72 OpenSIPS Control Panel (OCP) Installation Guide Good admin 19719   2014-08-13
 
71 Installation and configuration process record opensips opensips-cp admin 45649   2014-08-13
 
70 OpenSIPS as Homer Capture server admin 18869   2014-08-13
 
69 OpenSIPS , default script , Types of Routs , Routing in SIP, Video lecture admin 21010   2014-08-13
 
68 Configuracion de Kamailio 3.3 con NAT Traversal y XCAP. admin 21481   2014-08-12
 
67 Under RHEL6.5 install OpenSIPS 1.11.1 tls admin 20730   2014-08-12
 
66 OpenSIPS/OpenSER-a versatile SIP Server cfg admin 21651   2014-08-11
 
65 Kamailio Nat Traversal using RTPProxy admin 21233   2014-08-11
 
64 MediaProxy wiki page install configuration admin 21270   2014-08-11
 
63 오픈소스 (사내)메신저 서버 구축, 오픈 파이어(openfire) 설치방법과 세팅 admin 38495   2014-08-11
 
62 MediaProxy Installation Guide admin 20811   2014-08-10
 
61 RTPProxy 1.2.x Installation & Integration with OpenSIPS 1.5x admin 22031   2014-08-10
 
60 Opensips Installation, How to. Good guide wiki page admin 18989   2014-08-10
 
59 OpenSIPS Installation Notes admin 18523   2014-08-09
 
58 Installation and configuration process record opensips 1.9.1 admin 30834   2014-08-09
 
57 opensips 1.11.2 install Good Giide admin 21982   2014-08-09
 
56 fusionPBX install debian wheezy admin 21001   2014-08-09
 
55 opensips 1.11.2 install guide good 인스톨 가이드 admin 21262   2014-08-09
 
54 SigIMS IMS Platform admin 21595   2014-05-24
 
53 2013 2012년 분야별 최고의 오픈소스 소프트웨어 124선 admin 25662   2014-04-05
 
52 Video conference server OpenMCU-ru - Introduction admin 24169   2014-04-01
 
51 SIPSorcery admin 21989   2014-03-18
 
50 Ekiga (formely known as GnomeMeeting) is an open source SoftPhone admin 22406   2014-03-12
 
49 telepresence: Open Source SIP Telepresence/MCU admin 44022   2014-03-12
 
48 SIP PBX - OpenSIPS and Asterisk configuration admin 33488   2014-03-12
 
47 Conference Support in Kamailio (OpenSER) admin 28820   2014-03-12
 
46 OpenSIPS configuration for 2 or more FreeSWITCH installs admin 20470   2014-03-12
 
45 The Impact of TLS on SIP Server Performance file admin 22072   2014-03-12
 
44 book-opensips-101 / content / 3.2. SIP TLS Secure Calling.mediawiki admin 21300   2014-03-12
 
43 Where to check OpenSIPS does not start? admin 21388   2014-03-09
 
42 opensips-1.10.0_src.tar.gz experimental source code documentation admin 22409   2014-03-09
 
41 Kamailo OpenSIPs installation on Debian admin 27240   2014-03-09
 
40 Using the openSIPS Registrant Module admin 22838   2014-03-09
 
39 RTPproxy Frequentry Asked Questions (FAQ) ¶ admin 20817   2014-03-07
 
38 Building Telephony Systems with OpenSIPS 1.6 RTPProxy + OpenSIPS 1.7 admin 21946   2014-03-07
 
37 Installing RTPproxy Start RTPproxy in Bridged mode very good admin 34569   2014-03-07
 
36 OpenSIPS Control Panel (OCP) Installation Guide admin 20525   2014-03-06
 
35 OpenSIPS Control Panel install guide admin 21716   2014-03-06
 
34 rtpproxy Module admin 21774   2014-03-06
 
33 MediaProxy Installation Guide admin 29221   2014-03-06
 
32 How to install OpenSIPS on CentOS debian module add xcap admin 22581   2014-03-06
 
31 Problem with presence_xml module Opensips 1.9 admin 22092   2014-03-06
 
30 Building Telephony Systems with OpenSIPS 1.6 books file admin 23078   2014-03-06
 
29 Multimedia Service Platform admin 21423   2014-03-06
 
28 How to install OpenSIPS on CentOS Debian etc admin 22250   2014-03-05
 
27 Opensips Installation, How to. admin 18847   2014-03-05
 
26 100% CPU usage opensips admin 21619   2014-03-05
 
25 A2Billing and OpenSIPS admin 22871   2014-03-04
 
24 Opensips_1.9 install guide this is great I like this admin 28739   2014-03-04
 
23 Opensips install debian admin 22692   2014-03-03
 
22 Open Source VOIP applications, both clients and servers. admin 23134   2013-11-20
 
21 OfficeSIP Server is freeware VoIP, SIP server for Windows admin 24290   2013-09-11
 
20 My new toy: Bluebox-ng admin 38486   2013-04-06
 
19 Flooding Asterisk, Freeswitch and Kamailio with Metasploit admin 40080   2013-04-06
 
18 Asterisk Installation Asterisk Realtime configuration admin 27112   2013-04-06
 
17 The SIP Router Project admin 26119   2013-04-06
 
16 Kamailio :: A Quick Introduction admin 23527   2013-04-06
 
15 Welcome to the Smartvox Knowledgebase admin 23881   2013-04-06
 
14 Kamailio 3.3.x and Asterisk 10.7.0 Realtime Integration using Asterisk Database admin 28721   2013-04-06
 
13 OpenSIPS vs Asterisk admin 69684   2013-04-06