Distributed IP-PBX
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The document proposes an architecture for establishing a distributed IP-PBX communication system using multiple voice registers on different platforms and integrating both packet-switched and circuit-switched networks. It provides background on telecommunication technologies and protocols as well as an example case study of implementing the proposed architecture for a nationwide organization with distributed regional offices connected over an IP network. The case study demonstrates configuration of an Asterisk server and Cisco routers to enable voice communication between the regional branches using both the IP network and public switched telephone network.
![CASE STUDY: Asterisk Server Configuration
vi /etc/asterisk/extensions.conf
[cme-trunk]
Exten => _X.,1,Set(do_Voicemail=no)
Exten => _X.,n,NoOp(${CALLERID(num)})
Exten => _X.,n,Dial(SIP/${EXTEN})
Exten => _X.,n,NoOp(${HANGUPCAUSE} DAN ${DIALSTATUS})
Exten => _X.,n,Hangup()
;###Router01###
Exten => _20XX.,1,Set(do_Voicemail=no)
Exten => _20XX.,n,NoOp(${CALLERID(num)})
Exten => _20XX.,n,Dial(SIP/Router01/${EXTEN},,tTw)
Exten => _20XX.,n,NoOp(${HANGUPCAUSE} DAN ${DIALSTATUS})
Exten => _20XX.,n,Hangup()
;###Router02###
Exten => _21XX.,1,Set(do_Voicemail=no)
Exten => _21XX.,n,NoOp(${CALLERID(num)})
Exten => _21XX.,n,Dial(SIP/Router02/${EXTEN},,tTw)
Exten => _21XX.,n,NoOp(${HANGUPCAUSE} DAN ${DIALSTATUS})
Exten => _21XX.,n,Hangup()
vi /etc/asterisk/sip.conf
[Router01]
type=friend
host=10.11.121.2
dtmfmode=rfc2833
relaxdtmf=yes
canreinvite=no
insecure=port,invite
context=cme-trunk
quality=yes
nat=yes
Disallow=all
Allow=ulaw
Allow=alaw
[Router02]
type=friend
host=10.11.121.6
dtmfmode=rfc2833
relaxdtmf=yes
canreinvite=no
insecure=port,invite
context=cme-trunk
quality=yes
nat=yes
Disallow=all
Allow=ulaw
Allow=alaw
Context “cme-trunk”actually allows the router peer configurations at
sip.conf file. As for router01 information in sip.conf it gives informations for
peering router like IP, codec etc. As when the dial pattern needed the use
each other simultaneously. As for router01 dial pattern you can find that it
is seeking Router01 name from sip.conf](https://image.slidesharecdn.com/distributed-ip-pbx-151103173836-lva1-app6891/85/Distributed-IP-PBX-33-320.jpg)
Distributed IP-PBX
- 1. Distributed IP-PBX An Approach in Establishing Distributed communication using Multi Platform Voice Registers & integration of Packet & Circuit switched communication
- 2. AL FARUQ IBNA NAZIM Link3 Technologies Ltd. www.link3.net alfaruq@link3.net Contributor Acknowledgement Ahmed Sobhan – Link3 Technologies Ltd. Adnan Howlader – Bangladesh Microtechnology Ltd.
- 3. AGENDA • Background • Application and Appliances • Architecture & Benefits • Case Study • Key Findings
- 4. BACKGROUND
- 5. BACKGROUND : Telecommunication • A huge mix of evolved technologies. • Divided in circuit and packet switched network. Visualization is from the Opte Project of the various routes through a portion of the Internet.
- 6. BACKGROUND : Circuit Vs Packet switching
- 7. BACKGROUND : PABX, PSTN & IP-PBX • PBX: Switchboard operator managed system using cord circuits. • PABX (Private Automatic Branch Exchange): Availability of electromechanical switches gradually replaced manual switchboard – PBX. • PSTN (Public Switched Telephone Network): Aggregates circuit- switched telephone networks that are operated by national, regional, or local telephony operators. • IP PBX: Handles voice signals over Internet protocol, bringing benefits for computer telephony integration (CTI).
- 8. BACKGROUND: PSTN Circuits Core network is fully digitized with two variants • Analog (Rarely used now a days). • ISDN (Widely used in various aspect of communication network).
- 9. BACKGROUND: ISDN (Integrated Services for Digital Network) ISDN is a set of communication standards for simultaneous digital transmission of voice, video, data, and other network services over the traditional circuits of the public switched telephone network. • Basic Rate Interface (BRI) The entry level interface (BRI), a 128 kbit/s service delivered over a pair of standard telephone copper wires. The 144 kbit/s payload rate is broken down into two 64 kbit/s bearer channels ('B' channels) and one 16 kbit/s signalling channel ('D' channel or data channel). This is sometimes referred to as 2B+D. • Primary Rate Interface (PRI) PRI has an E1 (2048 kbit/s) which is carried in most part of the world. An E1 is 30 'B' channels of 64 kbit/s, one 'D' channel of 64 kbit/s and a timing and alarm channel of 64 kbit/s. In North America it is T1 (1544 kbit/s). It has 23 ‘B’ channels & 1 ‘D’ channels. Japan uses J1 which similar to T1. • Narrowband ISDN (N-ISDN) N-ISDN was introduce to replace analog to digital telephone system. It used a 64 kbit/s channel. Due to longer time to process standardization and growth and demand of capacity its was turned obsolete. • Broadband ISDN (B-ISDN) B-ISDN is considered using a better capacity bandwidth that is more than 1.544 Mbps. It is widely provided to use ATM at subscriber ends for better capacity. We can consider STM1 to STM4 in its genre. Later on the term broadband is taken to use for larger capable network capacity use for internet interfaces.
- 10. BACKGROUND: Codecs Codecs Payload Bitrate G.711 64 kbit/s G.726 16, 24 or 32 kbit/s G.723.1 5.3 or 6.3 kbit/s G.729 8 kbit/s GSM 13 kbit/s iLBC 13.3 or 15.2 kbit/s Speex 2.15 to 22.4 kbit/s G.711 is freely available as well gives highest quality. GSM is very popular due to good CPU and bandwidth tradeoff. iLBC & Speex is commonly used for voice transport over internet.
- 11. BACKGROUND: Signaling Protocol Signalling protocol is used to identify the state of connection between telephones or VOIP terminals (IP telephone or PCs or VoWLAN units). Common protocols: • H.323 • MGCP (Media Gateway Control Protocol) • SCCP (Skinny Client Control Protocol) • IAX (Inter Asterisk eXchange) • SIP (Session Initiation Protocol)
- 13. BACKGROUND: MGCP
- 14. BACKGROUND: SCCP & IAX SCCP • Skinny Client Control Protocol is a Cisco proprietary network terminal control protocol. • Lightweight protocol for session signalling with Cisco Unified Communications Manager/CallManager. IAX • Inter-Asterisk eXchange is native to the Asterisk PBX and few other soft switches. • Used for transporting VoIP telephony sessions between servers and terminal devices.
- 15. BACKGROUND: SIP (Session Initiation Protocol) • A signalling communications protocol. • Widely used for controlling multimedia communication sessions such as voice and video over IP.
- 17. APPLICATION & APPLIANCES : Contents Asterisk Cisco voice enabled routers Traditional phone sets IP Phone PABX stations
- 18. APPLICATION & APPLIANCES : Asterisk • A software implementation of a telephone private branch exchange (PBX). • Allows attached telephones to make calls to one another. • Connects other telephone services, such as the PSTN and VoIP services through media gateway. For features and details visit http://www.asterisk.org/get-started/features
- 19. APPLICATION & APPLIANCES : Cisco voice enabled routers Integrated services • Routing • PBX
- 20. APPLICATION & APPLIANCES : Traditional phone, IP-phone & PABX station
- 22. Considering Scenarios • Nationwide distributed work areas. • Implemented legacy PABX for internal communication. • Data & Internet connectivity. • Single box solution. • Area with lower resources considering rural area and environment.
- 26. Regional Zone 2 Branch 1 Regional Zone 2 Branch 2 Regional Zone 2 Branch 3 Regional Zone 3 Branch 1 Regional Zone 3 Branch 2 Regional Zone 3 Branch 3 Regional Zone 4 Branch 1 Regional Zone 4 Branch 2 Regional Zone 4 Branch 3 Regional Zone 5 Branch 1 Regional Zone 5 Branch 2 Regional Zone 5 Branch 3 Regional Zone 6 Branch 1 Regional Zone 6 Branch 2 Regional Zone 6 Branch 3 Regional Zone 1 Branch 1 Regional Zone 1 Branch 2 Regional Zone 1 Branch 3 Central Zone Regional Zone 1 Regional Zone 6 Regional Zone 5 Regional Zone 2 Regional Zone 4 Regional Zone 3 CASE SCENARIO: A Nationwide Spread Organization 1XXX 3XXX 2XXX 40XX 41XX 42XX 43XX 50XX 51XX 52XX 53XX 60XX 61XX 62XX63XX 70XX 71XX 72XX 73XX 80XX 81XX 82XX 83XX 90XX 91XX 92XX 93XX
- 27. Data Network Provider Zone 02 Zone 01 WWW CASE SCENARIO: Actual Network Diagram
- 28. CASE SCENARIO: Zone 1 Network Diagram PSTN PABX NETWORK 1 PABX NETWORK 2 PABX TERMINAL 1 PABX TERMINAL 2 4000 4001 4401 44024201 42024301 4302 4002 FXS CO FXO 4040 PABX NETWORK FXS 4500 4501 4502 FAX 4510 PABX NETWORK FXS 4400 FAX 4410 COCO PABX TERMINAL PABX TERMINAL PABX NETWORK FXS 4100 CO PABX TERMINAL FAX 4110 4101 4102 PABX NETWORK FXS 4200 CO PABX TERMINAL FAX 4210 FXS 4300 PABX TERMINAL FAX 4310 CO PABX NETWORK
- 30. Why Such integration? 1. Using the customers existing circuit setup PABX or PSTN. 2. Using cross site communication. 3. Using packet communication for connectivity. 4. Going ahead in advanced communication technology. 5. Cost benefit while transforming technology.
- 31. CASE STUDY
- 32. CASE STUDY: A sample Lab for such scenario 20122011 2010 2110 1010 Asterisk Server Core Switch Core Router Network Cloud Zone Router Distribution Switch Branch Router 1 Branch Switch Branch Router 2 Branch Switch FXO - 029117890 FXO - 029117891 Prime Integrations: 1. Asterisk Server. 2. Cisco Routers. 3. Legacy PABX integration. 4. PSTN connectivity integration.
- 33. CASE STUDY: Asterisk Server Configuration vi /etc/asterisk/extensions.conf [cme-trunk] Exten => _X.,1,Set(do_Voicemail=no) Exten => _X.,n,NoOp(${CALLERID(num)}) Exten => _X.,n,Dial(SIP/${EXTEN}) Exten => _X.,n,NoOp(${HANGUPCAUSE} DAN ${DIALSTATUS}) Exten => _X.,n,Hangup() ;###Router01### Exten => _20XX.,1,Set(do_Voicemail=no) Exten => _20XX.,n,NoOp(${CALLERID(num)}) Exten => _20XX.,n,Dial(SIP/Router01/${EXTEN},,tTw) Exten => _20XX.,n,NoOp(${HANGUPCAUSE} DAN ${DIALSTATUS}) Exten => _20XX.,n,Hangup() ;###Router02### Exten => _21XX.,1,Set(do_Voicemail=no) Exten => _21XX.,n,NoOp(${CALLERID(num)}) Exten => _21XX.,n,Dial(SIP/Router02/${EXTEN},,tTw) Exten => _21XX.,n,NoOp(${HANGUPCAUSE} DAN ${DIALSTATUS}) Exten => _21XX.,n,Hangup() vi /etc/asterisk/sip.conf [Router01] type=friend host=10.11.121.2 dtmfmode=rfc2833 relaxdtmf=yes canreinvite=no insecure=port,invite context=cme-trunk quality=yes nat=yes Disallow=all Allow=ulaw Allow=alaw [Router02] type=friend host=10.11.121.6 dtmfmode=rfc2833 relaxdtmf=yes canreinvite=no insecure=port,invite context=cme-trunk quality=yes nat=yes Disallow=all Allow=ulaw Allow=alaw Context “cme-trunk”actually allows the router peer configurations at sip.conf file. As for router01 information in sip.conf it gives informations for peering router like IP, codec etc. As when the dial pattern needed the use each other simultaneously. As for router01 dial pattern you can find that it is seeking Router01 name from sip.conf
- 34. CASE STUDY: Cisco Router Configuration ( Basic Configuration for voice ) voice service voip // Declaring the voice service over which mode. In our case its IP. allow-connections h323 to h323 allow-connections h323 to sip allow-connections sip to h323 allow-connections sip to sip supplementary-service h450.12 advertise-only // Common Information Additional Network Feature for H.323 fax protocol t38 ls-redundancy 0 hs-redundancy 0 fallback pass-through g711ulaw // Declaring fax protocol enabling low and high signal and giving option if fax protocol not available to go through an audio codec sip // sip configuration rel1xx disable // Reliable provisional response support disabled to stop error code registrar server expires max 1200 min 60 // Enabling SIP registry server and mentioning its expire time ! voice class codec 1 // declaring a codec group tag codec preference 1 g711ulaw codec preference 2 g711alaw ! voice register global // Global registry information declaration mode cme // mode defining to CME of Cisco source-address 10.11.121.2 port 5060 // Defining a registry server IP and port max-dn 10 // maximum dial no defining max-pool 10 // maximum pool defining tftp-path flash: // configuration loaded from flash with tftp create profile sync 0004028852090405 // Creating profile for IP phones ! ccm-manager application redundant port 5060 // Call manager application redundant port declaration ! dspfarm profile 1 transcode universal // Digital Signal Processor (DSP) profile for codec transformation for IP to IP media gateway description Transcoding codec g711ulaw codec g711alaw maximum sessions 5 associate application SCCP ! sip-ua // SIP user agent informations registrar ipv4:10.11.121.2 expires 3600 sip-server ipv4:10.11.121.2 ! telephony-service // CUCME configuration for router max-ephones 10 max-dn 10 system message #Router01# time-zone 21 max-conferences 8 gain -6 transfer-system full-consult !
- 35. CASE STUDY: Cisco Router Configuration ( Registering IP Phone / Pots Phone / Voice peer ) voice register dn 1 // Dial number declaration number 2001 allow watch name 2001 ! voice register pool 1 // Pool profile information for the number id mac 0030.4F7B.E3F9 number 1 dn 1 dtmf-relay sip-notify username 2001 password 123456 codec g711ulaw no vad ! dial-peer voice 1 pots // POTS number declaration destination-pattern 2002 incoming called-number .% port 0/0/0 ! dial-peer voice 2 voip // Peering with asterisk server description Router01-Asterisk destination-pattern 1... session protocol sipv2 session target ipv4:10.11.120.100:5060 dtmf-relay sip-notify codec g711ulaw no vad ! dial-peer voice 3 voip // Peering with routers description Router01-Router02 destination-pattern 21.. session protocol sipv2 session target ipv4:10.11.121.6:5060 dtmf-relay sip-notify codec g711ulaw no vad
- 36. CASE STUDY: Legacy PABX integration ! dial-peer voice 1 pots preference 1 destination-pattern 2000 incoming called-number .% port 0/0/0 ! dial-peer voice 2 pots preference 2 destination-pattern 2000 incoming called-number .% port 0/0/1 ! dial-peer voice 3 pots preference 3 destination-pattern 2000 incoming called-number .% port 0/0/2 ! dial-peer voice 4 pots preference 4 destination-pattern 2000 incoming called-number .% port 0/0/3 !
- 37. CASE STUDY: PSTN Connectivity Integration ( Call in/out for PSTN if considered legacy PBX) ! voice-port 0/3/1 connection plar opx 2000 description PSTN-FXO-Router1-PABX 029117891 caller-id enable ! voice-port 0/3/2 connection plar opx 2000 description PSTN-FXO-Router1-PABX 029117892 caller-id enable ! ! dial-peer voice 1 pots description PSTN-Out destination-pattern 0T direct-inward-dial forward-digits all port 0/3/1 ! dial-peer voice 2 pots description PSTN-Out destination-pattern 0T direct-inward-dial forward-digits all port 0/3/2 ! An incoming call to the PSTN numbers are forwarded to 2000 Call out numbers beginning with 0 Through PSTN ports
- 38. CASE STUDY: PSTN Connectivity Integration ( Call in/out for PSTN if considered legacy PBX) ! voice-port 0/3/1 connection plar opx 2000 description PSTN-FXO-Router1-PABX 029117891 caller-id enable ! voice-port 0/3/2 connection plar opx 2000 description PSTN-FXO-Router1-PABX 029117892 caller-id enable ! ! dial-peer voice 1 pots description PSTN-Out destination-pattern 0T direct-inward-dial forward-digits all port 0/3/1 ! dial-peer voice 2 pots description PSTN-Out destination-pattern 0T direct-inward-dial forward-digits all port 0/3/2 ! This will create a loop circuit for which all ports will be off-hook. This will allow all to use outgoing through PSTN
- 39. CASE STUDY: PSTN Connectivity Integration ( Call in for PSTN if considered legacy PBX) In such case its better if we use hunt group for PABX integration ! voice hunt-group 1 sequential list 2097, 2098, 2099 timeout 15 pilot 2000 ! dial-peer voice 1 pots destination-pattern 2097 incoming called-number .% port 0/0/0 ! dial-peer voice 2 pots destination-pattern 2098 incoming called-number .% port 0/0/1 ! dial-peer voice 3 pots destination-pattern 2099 incoming called-number .% port 0/0/2 ! 15 sec delay if not picked 15 sec delay if not picked 2097 2098 2099 2000
- 40. CASE STUDY: PSTN Connectivity Integration ( Dedicated PSTN and using it for remote router) Call in for Router1 ( Resided in Router2) ! voice-port 0/3/1 connection plar opx 2010 description PSTN-FXO-Router1-PABX 029117891 caller-id enable ! voice-port 0/3/2 connection plar opx 2012 description PSTN-FXO-Router1-PABX 029117892 caller-id enable ! Call out for Router1 ( Resided in Router1) ! voice translation-rule 1 rule 1 /^0/ /A20100/ ! voice translation-rule 2 rule 1 /^0/ /A20120/ ! voice translation-profile phone2010 translate called 1 ! voice translation-profile phone2012 translate called 2 ! voice register dn 1 translation-profile incoming phone2010 number 2010 allow watch name 2010 ! ! dial-peer voice 1 pots translation-profile incoming phone2012 destination-pattern 2012 incoming called-number .% port 0/0/0 ! dial-peer voice 2 voip description PSTN-OUT-2010 destination-pattern A2010T session protocol sipv2 session target ipv4:10.11.121.2 dtmf-relay sip-notify codec g711ulaw no vad ! dial-peer voice 3 voip description PSTN-OUT-2012 destination-pattern A2012T session protocol sipv2 session target ipv4:10.11.121.2 dtmf-relay sip-notify codec g711ulaw no vad ! Call out for Router1 ( Resided in Router2) dial-peer voice 1 pots description Call-Out-For-2010 destination-pattern A2010T direct-inward-dial port 0/3/0 ! dial-peer voice 2 pots description Call-Out-For-2012 destination-pattern A4202T direct-inward-dial port 0/3/1
- 41. CASE STUDY: PSTN Connectivity Integration ( Dedicated PSTN and using it for remote router) Call in for Router1 ( Resided in Router2) ! voice-port 0/3/1 connection plar opx 2010 description PSTN-FXO-Router1-PABX 029117891 caller-id enable ! Call out for Router1 ( Resided in Router1) ! voice translation-rule 1 rule 1 /^0/ /A20100/ ! voice translation-profile phone2010 translate called 1 ! dial-peer voice 1 pots translation-profile incoming phone2012 destination-pattern 2012 incoming called-number .% port 0/0/0 ! dial-peer voice 2 voip description PSTN-OUT-2010 destination-pattern A2010T session protocol sipv2 session target ipv4:10.11.121.2 dtmf-relay sip-notify codec g711ulaw no vad ! ! Call out for Router1 ( Resided in Router2) ! dial-peer voice 1 pots description Call-Out-For-2010 destination-pattern A2010T direct-inward-dial port 0/3/0 ! Router 2 Router 1 PSTN 2010 A call coming to 029117891 2010 dials 01711234567 and that transforms in A201001711234567 which is mentioned as A2010T to transport to Router 2 As the router gets A2010T where T = 01711234567 It will only forward T to its mentioned port of PSTN Call going to 01711234567
- 42. CASE STUDY: Remote FXO activity due to local device or data network failure ! voice-port 0/3/1 supervisory disconnect dualtone mid-call cptone GB timeouts call-disconnect 5 timeouts wait-release 5 connection plar opx 2010 impedance complex2 description PSTN-FXO-Router1-PABX 029117891 caller-id enable ! voice-port 0/3/2 supervisory disconnect dualtone mid-call cptone GB timeouts call-disconnect 5 timeouts wait-release 5 connection plar opx 2012 impedance complex2 description PSTN-FXO-Router1-PABX 029117891 caller-id enable ! The FXO port will be offhook and no connection can be established other than port reset. So the following should be maintained for FXO ports.
- 43. Conclusion: Final Achievements and Future 1. A distributed organization can operate on their own communication pattern. 2. While merging to this technology their existing legacy technology PABX is integrated. 3. Zone based registry allows you to have you own area communication due to unavailability of data connection. 4. Local or remote PSTN integration. Future 1. FAX integration or personal FAX network. 2. Achieving a single area of communication management for voice, video, data, internet etc. 3. Achieving IP-PABX smart features. 4. Using data communication more efficiently sharing it with QOS. 5. Getting IP-TSP trunks as an alternative option of PSTN.