ALMATY UNIVERSITY OF POWER ENGINEERING AND TELECOMMUNICATIONS

АУЭС

Non-profit joint-stock company

The chair of infocommunicationaltechnology

BASICS OF IP – TELEPHONY

Guidelines for laboratory works5В071900 specialty – Radio engineering, electronics and telecommunications

Almaty 2016

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Summary plan 2015, pos. 123

Katipa ChezhimbayevaSharafat Mirzakulova

Vyacheslav Bublik

BASICS OF IP - TELEPHONY

Guidelines for laboratory works5В071900 specialty – Radio engineering, electronics and telecommunications

Editor Kozlov V. S.Specialist for standardization Moldabekova N. K.

Approved for publishing __ . __ . ____ Format 60х84 1/16Release 20 items Paper typographic №1Value 2 ed.-pub.p. Order___ Price 1000 tenge

Copy office ofnon-profit joint-stock company

"Almaty university of power engineering and telecommunications"050013, Almaty, Baitursynov str., 126

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Contents

Introduction . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 41 Laboratory work №1. Register SIP-softphone in FreeSWITCH . . . . . . . . . . . . . . 52 Laboratory work №2. Creating of new SIP account in FreeSWITCH . . . . . . . . . 113 Laboratory work №3. Creating Entries in the numbering plan of FreeSWITCH . 144 Laboratory work №4. Video calls and audio conference in FreeSWITCH . . . . . 165 Laboratory work №5. Configuring of SIP-users for Asterisk software . . . . . . . . 206 Laboratory work №6. Configuration of ISDN PRI trunk between two Asteriskservers . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 247 Laboratory work №7. Configuring of SIP trunk between two Asterisk servers . .268 Laboratory work №8. Configure SIP trunk reservation by ISDN PRI trunkbetween two Asterisk servers . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .28Abbreviations . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .30References . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 31

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Introduction

This student exercises guide on discipline "IP-telephony fundamentals"allows students to become familiar with the distinctive features of an IP-telephonyfrom the traditional telephony, as well as obtaining the basic skills in configuring ofthe IP-based telephony using software and hardware:

- FreeSWITCH software;- Asterisk software;- VRX-1010-E1 gateway;- Digium TE110P;- 3CX Phone IP-phone software;- D-Link DES-1024 switch ;- multimedia consoles. The development of information technology and telecommunications has a

decisive impact on the society and economy. Under their influence is increasingimportance of information as a market product, radically changing the way peoplelive.

Transition to IP-based networks with packet switched telephone network isinevitable with regard to its efficiency as compared to traditional networks. Morefunctional IP-PBX (PBX), with support for IP-telephony both in hardware andsoftware design (virtual) came in place of the classic digital ATS.

Tendencies of development of IP-PBX features (including virtual) are suchthat in the near future function of speech synthesis and voice recognition be will bewidely used, further penetration of unified communications, demand for recordingand storing of all transmitted data (video call recording) and others are expected..That is, in the future, IP-PBX - is a unified communications center automation of acompany.

Eight new exercises were developed based on PBX FreeSWITCH (privatebranch telephone exchange software running on the IP protocol).

IP-telephony means voice that carried over data networks, in particular IP-based networks (IP - Internet Protocol). When using an IP-telephony data packetspass through a local or wide area network, and they have a certain IP-address, fromwhich it is transmitted.

VoIP-Gateway (Voice over IP-gateway) - a device designed to connectphones or PBX to the IP-based networks to transmit voice traffic through it.

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1 Laboratory work №1. Registration of SIP-softphone in FreeSWITCHsoftware

ObjectiveThe purposes of this exercise are to introduce the software, to obtain the basic

skills of working with configuration files of FreeSWITCH software and softphone3CX Phone.

1.1 Preparation

Familiarization with the basics of IP-telephony technology, with the mainsoftware functionality of FreeSWITCH software and softphone 3CX Phone.

1.2 Laboratory work task

1.2.1 Check your SIP account on FreeSWITCH server in accordance with anassignment.

1.2.2 Configure your softphone 3CX Phone account in accordance with thespecified option.

1.2.3 Make multiple voice calls to verify the settings of softphone.1.2.4 Configure the programmable buttons of 3CX Phone softphone in

accordance with the specified option to test it.1.2.5 Create a progress report.

1.3 Guidelines

1.3.1 FreeSWITCH system was created in 2006 by Anthony Minessale, oneof the former developers of Asterisk. In developing the architecture ofFreeSWITCH the existing problems of the time of open software products for IP-telephony were taken into account. Therefore, the novelty is stable and is able towork not only on Linux, but on Windows. The main interface to configureFreeSWITCH are text files in XML format.

FreeSWITCH system can be used as a switch, PBX, a media gateway or amedia server for IVR applications, which uses simple or XML scripts forcontrolling the call processing algorithm. FreeSWITCH supports multiple protocols,such as SIP, H.323, IAX2 and Google Talk, which allows you to interact with sipX,OpenPBX, Bayonne, YATE, or Asterisk. FreeSWITCH supports many advancedfeatures of SIP, such as the presence/BLF/SLA, TCP TLS and sRTP others. Voicechannels and conferences can operate at frequencies of 8, 16, 32 and 48 kHz, andallow you to combine the channels with different frequencies.

FreeSWITCH supports a wide range of functions, from the voice menu, callstatistics, to voice mail, call-center etc. Among its main advantages, highlighted are:

- rich functionality;- support a variety of telephony interfaces: analog, digital, IP telephony;

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- simple connection of IP phones and high-quality communications.1.3.2 Run WinSCP to connect a computer to a remote server. WinSCP allows

connection to the server using SSH protocol, SFTP (SSH File Trasfer Protocol) andSCP (Secure Copy Protocol), as a rule, with the machines under the UNIXoperating system.

In the window that appears, fill in the following fields:File protocol: SFTPHost name: 192.168.1.67Port number: 22User name: rootPassword: qscaxz1.3.3 Confirm by pressing the Login (figure 1.1).

Figure 1.1 –WinSCP connection window

1.3.4 In the File Manager window (figure 1.2) the right pane displays the fileson the remote server, and the left - the files on the local computer.

1.3.5 On the remote server, change to the directory with SIP accounts/usr/local/freeswitch/conf/directory/default (figure 1.3).

1.3.6 Make sure that the file that corresponds to your option, exists in thespecified directory. For example, if an option of the account is 1000, the version ofthe file corresponds to 1000.xml.

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Figure 1.2 –Main window of WinSCP

Figure 1.3 –Contents of directory /usr/local/freeswitch/conf/directory/default

1.3.7 Run the 3CX Phone. In the window that appears (figure 1.4) press MainMenu.

1.3.8 On the panel that appears (figure 1.5), click Accounts.1.3.9 In the next window (figure 1.6), click the New.

Figure 1.4 –Main window of 3CX Phone before registration on server

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Figure 1.5 –Main menu window of3CX Phone

Figure 1.6 –Window of accounts of 3CX Phone

1.3.9 In the next window (figure 1.7), specify the following:Account name: Student_XXCaller ID: YYYYExtension: YYYYID: YYYYPassword: 1234local IP 192.168.1.67Where: XX – option number; YYYY – account according to option number.

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Figure 1.7 –Account settings window of3CX Phone

1.3.10 Close the window (figure 1.7, figure 1.6) by pressing the OK.1.3.11 The status bar should display Hook (figure 1.8).

Figure 1.8 –Main window of3CX Phone after registration on server

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1.3.12 Make voice calls to several numbers in the range of 1000 to 1019.Make sure that the calls are established.

1.3.13 Open the programmable buttons panel. Right click on the firstprogrammable button.

1.3.14 In the window (figure 1.9) fill in the Label and User ID. Close thewindow by pressing OK.

1.3.15 Test the programmed button.

Figure 1.9 –Configuration window of programmable button

1.3.16 Create a report on the work done. Input data are given in table 1.1.

Table 1.1 — Input data№ Account № Account № Account № Account1 1000 6 1005 11 1010 16 10152 1001 7 1006 12 1011 17 10163 1002 8 1007 13 1012 18 10174 1003 9 1008 14 1013 19 10185 1004 10 1009 15 1014 20 1019

1.4 Quiz

1.4.1 What are the analogue of FreeSWITCH?1.4.2 What are the analogue of3CX Phone software? 1.4.3 What VoIP services of telecommunication providers you know?1.4.4 What information is available to the client from provider to configure

IP-Phone?

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2 Laboratory work №2. Creating new SIP account inFreeSWITCH

ObjectiveThe purpose of this exercise is to acquiring skills of editing SIP accounts

configuration files of FreeSWITCH software.

2.1 Preparation

Introduction to the SIP protocol and to its principles, to configuration files ofFreeSWITCH software, which contains settings of the SIP protocol.

2.2 Laboratory work task

2.2.1 Create SIP account on FreeSWITCH server in accordance with anassignment.

2.2.2 Configure account of 3CX Phone softphone in accordance with thecreated SIP account.

2.2.3 Make multiple voice calls to verify the configuration of software phoneand server.

2.2.4 Create a progress report.

2.3 Guidelines

2.3.1 Start Notepad. Enter the text in the window based on a template (figure2.1). Make changes to the template in accordance with the SIP account dataaccording to an option.

Figure 2.1 –The template for the SIP account with the ID 1012

2.3.2 Save the text in the menu File - Save As ... (see figure 2.2).

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Figure 2.2 –SIP account saving

2.3.3 In the window (figure 2.3) in the File name (file name) enter YYYY.xml,where YYYY - SIP account ID according to an option. In the Save as type (file type),select All files (All files).

2.3.4 Click Save.2.3.5 Run WinSCP and connect to the server (exercise №1).2.3.6 On the remote side, specify the directory

/usr/local/freeswitch/conf/directory/default.

Figure 2.3 –Select the type and file name extension

2.3.7 On the local side, specify the folder that contains the previously savedfile.

2.3.8 On the local side, select the previously saved file. Then press the F5 keyor click F5 Copy button (figure 2.4).

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Figure 2.4 –File transfer

2.3.9 Tell the teacher about the result of the file transfer.2.3.10 Run the 3CX Phone and configure account, using the data from the file

created exercise №1). Make sure registration on the server is successful.2.3.11 Make voice calls to several numbers in the range of 1000-1019. Make

sure that the calls are established.2.3.12 Create a report on the work done. Input data are given in Table 2.1.

Table 2.1 –Input data№ SIP account Password № SIP account Password1 1020 qwe 11 1030 edc2 1021 rty 12 1031 rfv3 1022 uio 13 1032 tgb4 1023 asd 14 1033 yhn5 1024 fgh 15 1034 ujm6 1025 jkl 16 1035 qaw7 1026 zxc 17 1036 sed8 1027 vbn 18 1037 rft9 1028 qaz 19 1038 gyh10 1029 wsx 20 1039 uji

2.4 Quiz

2.4.1 In what format FreeSWITCH software configuration files are stored?2.4.2 What standard is used in the structuring of configuration files of

FreeSWITCH software?2.4.3 Which directories contain configuration files of FreeSWITCH?2.4.4 What programs needed to change the configuration files of

FreeSWITCH?

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3 Laboratory work №3. Creating Entries in the numbering planof FreeSWITCH

ObjectiveThe purpose of this exercise is to obtain skills in editing configuration files of

SIP numbering plan of freeswitch.

3.1 Preparation

Familiarization with the basics of numbering in telephone networks, with theassigning of numbering plans and with configuration files of FreeSWITCH, whichcontains the settings of the numbering plan.

3.2 Laboratory work task

3.2.1 Create a numbering plan entry for SIP account in FreeSWITCH, createdin the previous exercise according to a given option.

3.2.2 Load configuration files to FreeSWITCH.3.2.3 Make multiple voice calls to verify the configuration of the numbering

plan.3.2.4 Create a report.

3.3 Guidelines

3.3.1 Start Notepad. Enter the text based on a template in the window (figure3.1). Make changes to the template in accordance with the entry of numbering planaccording to an option.

Figure 3.1 –Template entry of numbering plan with the number of conformity 1012for SIP account with the ID 1012

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3.3.2 Save the file as YYYYexten.xml on the local computer (Exercise №2).Where YYYY - number of conformity in the entry of the numbering plan accordingto an option.

3.3.3 Copy created file to directory /usr/local/freeswitch/conf/dialplan/default(exercise№2).

3.3.4 Run PuTTY. In session configuration window PuTTY (figure 3.2) inHost Name (or IP address) set 192.168.1.67. Remaining form keep unchanged.Press Open button.

3.3.5 In PuTTY session window (figure 3.3) enter login name root andpassword qscaxz.

3.3.6 After the appearance of the invitation symbol # enter service freeswitchrestart command for reload of settings of FreeSWITCH.

Figure 3.2 – PuTTY Session configuration window

Figure 3.3 – PuTTY session window

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3.3.7 Run 3CX Phone and configure the account, using the data from the filecreated when performing exercise №2 (exercises №1,2). Make sure to checkregistration on the server is successful.

3.3.8 Make voice calls to several numbers in the range of 1020 - 1039. Makesure that the calls are established.

3.3.9 Create a report. Initial data in table 3.1.

Table 3.1 –Initial data№ Numbering plan

numberSIPaccountID

№ Number ofnumbering plan

SIPaccountID

1 1020 1020 11 1030 10302 1021 1021 12 1031 10313 1022 1022 13 1032 10324 1023 1023 14 1033 10335 1024 1024 15 1034 10346 1025 1025 16 1035 10357 1026 1026 17 1036 10368 1027 1027 18 1037 10379 1028 1028 19 1038 103810 1029 1029 20 1039 1039

3.4 Quiz

3.4.1 The purpose of numbering plan?3.4.2 What actions can be mapped to the numbers in the numbering plan of

FreeSWITCH software?3.4.3 The purpose of PuTTY?3.4.4 The purpose of service freeswitch reload?

4 Laboratory work №4. Video calls and audio conferencein FreeSWITCH

ObjectiveThe purpose of this exercise is to obtain skills in using softphone 3CX Phone

and FreeSWITCH for video calls and audio conferencing.

4.1 Preparation

Familiarization with the codecs used for video calling, and theircharacteristics, the basics of audio conferencing and editing configuration files ofFreeSWITCH, which contain settings of audio conferencing.

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4.2 Laboratory work task

4.2.1 Make a few video calls between with different codecs settingsaccording to the option.

4.2.2 Make a connection with preconfigured audio conference, in accordancewith a given option.

4.2.3 Compare the quality of the sound in a different of audio conferences.4.2.4 Create a report.

4.3 Guidelines

4.3.1 Run the 3CX Phone and configure account, using the data from the fileyou created in Exercise 2 (exercise №1,2). Make sure to check the registration onthe server is successful.

4.3.2 Open the account settings window of 3CX Phone (figure 1.7, exercise№1) for the caller. In this window, click Advanced settings.

4.3.3 In the next window (figure 4.1) in the section Video formats configurepriorities of video formats according to an option. Close the window by pressingOK.

4.3.4 Perform steps 4.3.2 - 4.3.3 for the called party.4.3.5 Press open video panel button. Make sure the phone is connected to a

local web-camera (figure 4.2).

Figure 4.1 –Account advanced settings window of 3CX Phone

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Figure 4.2 –Main window of 3CX Phone with opened video panel

4.3.6 On the caller side dial the called party number and make a call. Makesure that you have the two-way audio/video connection (see figure 4.3). Write downused video format.

4.3.8 Change the order of priorities of video formats for caller and calledparty (figure 4.1) on your own. Re-make a video call. Write down used videoformat.

4.3.9 FreeSWITCH software has three pre-configured groups of staticconferences, with different codecs used:

- static conference with narrowband codecs. One hundred conferences havephone numbers from 3000 to 3099;

- static conference with broadband codecs. One hundred conferences havephone numbers from 3100 to 3199;

- static conference with ultra-broadband codecs. One hundred of conferenceshave phone numbers from 3200 to 3299.

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Figure 4.3 –Main window of 3CX Phone during video call

4.3.10 Make a voice call with two or more phone to static conferencesnumbers according to an option. Evaluate and compare the sound quality in theaffected static conferences.

4.3.9 Create a report. Input data are given in Table 4.1.

Table 4.1 –Initial data№ Video formats priorities of

callerVideo formats priorities of

called partyList of static

conference numbers

1 704 х 576; 352 х 288;176 х 144; 128 х 96

704 х 576; 352 х 288;176 х 144; 128 х 96

3001, 3101, 3201

2 704 х 576; 352 х 288;176 х 144; 128 х 96

352 х 288; 176 х 144;128 х 96; 704 х 576

3002, 3102, 3202

3 704 х 576; 352 х 288;176 х 144; 128 х 96

176 х 144; 128 х 96;704 х 576; 352 х 288

3003, 3103, 3203

4 704 х 576; 352 х 288;176 х 144; 128 х 96

128 х 96; 704 х 576;352 х 288; 176 х 144

3004, 3104, 3204

5 352 х 288; 176 х 144; 704 х 576; 352 х 288; 3005, 3105, 3205

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№ Video formats priorities ofcaller

Video formats priorities ofcalled party

List of staticconference numbers

128 х 96; 704 х 576 176 х 144; 128 х 96

6 352 х 288; 176 х 144; 128 х 96; 704 х 576

352 х 288; 176 х 144;128 х 96; 704 х 576

3006, 3106, 3206

7 352 х 288; 176 х 144;128 х 96; 704 х 576

176 х 144; 128 х 96704 х 576; 352 х 288

3007, 3107, 3207

8 352 х 288; 176 х 144;128 х 96; 704 х 576

128 х 96; 704 х 576;352 х 288; 176 х 144

3008, 3108, 3208

9 176 х 144; 128 х 96;704 х 576; 352 х 288

704 х 576; 352 х 288;176 х 144; 128 х 96

3009, 3109, 3209

10 176 х 144; 128 х 96;704 х 576; 352 х 288

352 х 288; 176 х 144;128 х 96; 704 х 576

3010, 3110, 3210

4.4 Quiz

4.4.1 Video quality during video call depends on what parameters ?4.4.2 Main codecs characteristics?4.4.3 Criteria for codecs selection?4.4.4 What do codecs and transcoding means ?

5 Laboratory work №5. Configuring of SIP-users for Asterisk software

ObjectiveThe purpose of this exercise is to provide basic skills in configuration of IP-

based network (Ethernet Switch of second level, Asterisk for Linux operatingsystem and 3CX Phone softphone), the configuration of the gateway VoIP, neededto communicate with the traditional telephone network.

5.1 Preparation

Familiarize with the IP-based network (the second level switch, gateway,3CX Phone softphone and Asterisk) and with the features and capabilities of theoperating system Linux (the design of the file system, directory structure, the majordistributions, basic commands for configuring services).

5.2 Laboratory work task

5.2.1 Build the circuit using a gateway VRX-1010-E1, the switch of thesecond level D-Link DES-1024, a PC with 3CX Phone installed.

5.2.2 Assign IP address and subnet mask for PC.5.2.3 Connect cable to the console port VRX-1010-E1 via SSH-client.

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5.2.4 Familiarize with the implementation of the various procedures andcommands.

5.2.5 Connect multimedia consoles to PC.5.2.6 Run the 3CX Phone on PC and configure to connect to the gateway.5.2.6 Make test calls between PCs.5.2.7 Create a report.

5.3 Guidelines

5.3.1 Asterisk - is a software PBX that can switch both VoIP calls and callsmade between IP-phones, the traditional telephone network. The number ofsubscribers in the network can reach 2000, and is limited only by capacity of theserver.

Asterisk supports such protocols as IAX, SIP, H.323, Skinny, UNIStim.Asterisk supports such codecs as G.711 (ulaw/alaw), G.722, G.723, G.729,

GSM, iLBC, LPC-10, Speex.The basic of the principles of universal gateway VRX-1010-E1 are the

flexibility and scalability of both software and hardware. VoIP-Gateway VRX-1010-E1 is connected to the analog and digital PBX

systems of almost any manufacturer (Panasonic, LG, Samsung, Siemens, etc..).Universal VoIP VRX-1010-E1 gateway is implemented as a complete

structural element. Gateway Controller is based on a powerful single-chiptelecommunication processor, in fact, it is an industrial computer, operating underthe control of the built-in Linux operating system.

Gateway management program is implemented as a service (daemon) of theoperating system Linux, which allows system control program to operate as adedicated server, and directly on the board of the universal gateway.

Gateway management software and software management system interactthrough the protocol stack TCP-IP, which allows you to build distributed systems ofany complexity. It supports the following protocols, control/alarm:

- support of SIP and H.323 protocols;- advanced functions of QoS;- secure functions (user authentication, access control lists);- transmit/receiveof faxes (FAX over IP);- web-GUI configuration.Universal Gateway VRX-1010-E1 - modular digital device that has a

powerful processor dedicated to broadcast 2 megabit streams of E1 of the branchexchange (PBX) to the SIP server (SIP PBX).

Universal VoIP gateway will combine into one transmission medium for datanetworks and traditional telephone network.

5.3.2 Connect network devices in accordance circuit shown in figure 5.1.

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Figure 5.1 – Network diagram

5.3.3 Assign IP-address and a subnet mask to a PC according to an option(Table 1.1). As the default gateway address, set the IP-gateway VRX-1010-E1.

5.3.4 Connect to the console gateway VRX-1010-E1 via SSH-client5.3.5 Bring the contents of the file to the form /etc/asterisk/asterisk.conf:[directories](!)astetcdir => /etc/asteriskastmoddir => /usr/lib64/asterisk/modulesastvarlibdir => /var/lib/asteriskastdbdir => /var/lib/asteriskastkeydir => /var/lib/asteriskastdatadir => /usr/share/asteriskastagidir => /usr/share/asterisk/agi-binastspooldir => /var/spool/asteriskastrundir => /run/asteriskastlogdir => /var/log/asterisk[options]languageprefix = yesrunuser = asterisk rungroup = asteriskdocumentation_language = en_US[compat]pbx_realtime=1.6res_agi=1.6app_set=1.6 5.3.6 Bring the contents of the file to the form /etc/asterisk/sip.conf[general]context=default allowoverlap=no udpenable=yesudpbindaddr=0.0.0.0 tcpenable=no srvlookup=yes

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videosupport=yes5.3.7 Bring the contents of the file to the form /etc/asterisk/extensions.conf[general]static=yeswriteprotect=noclearglobalvars=no[globals]CONSOLE=Console/dspIAXINFO=guest TRUNKMSD=1 [office]5.3.8 Add the following lines to the file /etc/asterisk/sip.conf[<X>]type=frienddtmfmode=rfc2833host=dynamicsecret=<pass>context=officeWhere <X> and <pass> - login name and password according to an option.5.3.9 Add the contents of the file /etc/asterisk/extensions.conf the following

lines exten =><X>,1,Dial(SIP/<X>)same =>n,HangUp()Where <X> - login name according to an option.5.3.10 In the console, enter the following commandsservice asterisk restartservice iptables stop5.3.11 On the PC, run the 3CX Phone, configure it and connect to the

gateway.5.3.12 Make test calls between PCs5.3.13 Create a report.

5.4Quiz

5.4.1 What are the analogues of application processes serving as a virtualPBX?

5.4.2 What are the analogues of3CX Phone? 5.4.3 What VoIP services of local telco providers do you know?5.4.4 What information is available to the client from provider to configure

IP-Phone?5.4.5 What transformations occur in the transmission of voice signals over IP-

based networks?

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Table 5.1 –Initial data№ IP address № IP address № IP address1 168.192.1.1 8 168.192.1.8 15 168.192.1.152 168.192.1.2 9 168.192.1.9 16 168.192.1.163 168.192.1.3 10 168.192.1.10 17 168.192.1.174 168.192.1.4 11 168.192.1.11 18 168.192.1.185 168.192.1.5 12 168.192.1.12 19 168.192.1.196 168.192.1.6 13 168.192.1.13 20 168.192.1.207 168.192.1.7 14 168.192.1.14

6 Laboratory work №6. Configuration of ISDN PRI trunkbetween two Asterisk servers

ObjectiveAsterisk The purpose of this exercise is to provide basic skills in working

with PRI interfaces and ISDN PRI trunk configuration between two Asteriskservers.

6.1 Preparation

6.1.1 Familiarization with the primary interface PRI, performing the role of aconnecting line between the two IP PBX Asterisk.

6.2 Laboratory work task

6.2.1 Build the circuit using gateways VRX-1010-E1, the switch of thesecond level D-Link DES-1024, a PC with 3CX Phone software.

6.2.2 Assign IP-addressandsubnet mask to PC.6.2.3 Connect network cable to console port of VRX-1010-E1 gatewayusing

SSH-client.6.2.4 Familiarize with the implementation of the various procedures and

commands.6.2.5 Connect multimedia console to PC.6.2.6 RunНа3CX Phone on PC andconfigure to connect to gateway.6.2.6 Make test calls between PCs via a connecting link based on PRI

interface.6.2.7 Create a report.

6. Guidelines

6.3.1 Modern PBXs have the following interfaces:- Z (wire);- BRI (main);- PRI (primary);

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- E1 (digital stream at 2048 kbps) for connection with another devices. For connection to a data network PRI interfaces and E1 can be used. Connect network devices according to circuit shown in figure 6.1.

Figure 6.1 –Network diagram

6.3.2 Assign IP-address and subnet mask to PC according to an option (Table1.1). As the default gateway address, set the gateway IP of VRX-1010-E1(configure according to exercise 1 for each gateway VRX-1010-E1).

6.3.3 Connect to console VRX-1010-E1 gateway via SSH-client. Enter nextcommands in console:

serviceiptables stopmodprobe wcte11xpdahdi_genconf6.3.4 Bring the contents of the file /etc/dahdi/system.conf to the following

formspan=1,1,0,ccs,hdb3,crc4bchan=1-15,17-31dchan=16loadzone=rudefaultzone=ru6.3.5 In console, enter following command service dahdi start6.3.6 Bring the contents of the file /etc/asterisk/chan_dahdi.conf to the

following formgroup=1context=officeechocancel=yesechocancelwhenbridged=noechotraining=yesswitchtype=euroisdnsignaling=pri_cpe ;pri_netchannel => 1-15,17-316.3.7 Add to the file /etc/asterisk/extensions.conffollowing linesexten => _9X.,1,Dial(DAHDI/g1/{EXTEN:1})same =>n,HangUp()6.3.8 In console, enter following commandsservice asterisk restart

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asterisk –r6.3.9 Check the status of DAHDI subsystem with following command dahdi

show status6.3.10 Make test calls between PCs on different gateways via prefix.6.3.11 Create a report.

Table 6.1 –Initial data№ SIP-address № SIP-address № SIP-address № SIP-address1 1000 6 1005 11 1010 16 10152 1001 7 1006 12 1011 17 10163 1002 8 1007 13 1012 18 10174 1003 9 1008 14 1013 19 10185 1004 10 1009 15 1014 20 1019

6.4 Quiz

6.4.1 What intefaces in digital network with integration of services exist? 6.4.2 What does BRI mean? 6.4.3 What does PRI mean?6.4.4 What does E1/T1 mean?6.4.5 What gateways do you know?6.4.6 Describe the process of data transmission in accordance with figure 2.1.

7 Laboratory work №7. Configuring of SIP trunk between two Asteriskservers

ObjectiveThe purpose of this exercise is to configure the SIP-trunk between two

Asterisk servers.

7.1 Preparation

7.1.1 Familiarization with the SIP protocol and its role in the transmission ofvoice signals.

7.2 Laboratory work task

7.2.1 Build the circuit using VRX-1010-E1 gateways, the switch of thesecond level D-Link DES-1024, 3CX Phone software for PC.

7.2.2 Assign IP address and subnet mask to PC.7.2.3 Connect to VRX-1010-E1 using network cable gateway and SSH-client.7.2.4 Familiarize with the implementation of the various procedures and

commands.7.2.5 Connect multimedia console ro PC.

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7.2.6 Start 3CX Phone on PC and configure it to connect to gateway.7.2.6 Make test calls between PCs via a link based on PRI interface.7.2.7 Create a report.

7.3 Guidelines

7.3.1 Build the circuit as shown in figure7.1.7.3.2 Configure according to exeercise5for every VRX-1010-E1 gateway.7.3.3 Add to the file /etc/asterisk/sip.conf following lines:[<X>]type=peerdtmfmode=rfc2833host=<remoteIP>secret=<pass>context=officeWhere <X> - name of remote gateway according to an option, <remoteIP> -

IP-address of remote gateway, <pass> - shared with the remote gateway password.

Figure 7.1 –Network diagram

7.3.4 Add to the file /etc/asterisk/extensions.conf following linesexten => _0X.,1,Dial(SIP/<X>/{EXTEN:1})same =>n,HangUp()Where <X> - remote gateway name according to an option.7.3.5 In console, enter the following command service asterisk restart7.3.6 Make test calls between PCs on different gateways, using prefix 0.7.3.7 Create a report.

Table 7.1 –Initial data№ SIP-address № SIP-address № SIP-address1 1000 GW1 8 1007 GW2 15 1014 GW12 1001 GW2 9 1008 GW1 16 1015 GW23 1002 GW1 10 1009 GW2 17 1016 GW14 1003 GW2 11 1010 GW1 18 1017 GW25 1004 GW1 12 1011 GW2 19 1018 GW16 1005 GW2 13 1012 GW1 20 1019 GW27 1006 GW1 14 1013 GW2

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7.4 Quiz

7.4.1 SIP protocol? 7.4.2 What are functions of SIP-server? 7.4.3 What are functions of gate keeper?7.4.4 What are functions of proxy-sever?7.4.5 What functions of location determination server?7.4.6 What are functions of forwarding server?7.4.7 Describe the process of data transmission according to figure 7.1.

8 Laboratory work №8. Configure SIP trunk reservation by ISDN PRItrunk between two Asterisk servers

ObjectiveThe purpose of this exercise is to configure the backup of SIP-trunk with

ISDN PRI trunk between two Asterisk servers.

8.1 Preparation

8.1.1 Familiarization with the configuration of backup SIP-trunk, ISDN PRItrunk in network with Asterisk software during voice signals calls.

8.2 Lab work task

8.2.1 Build the circuit using VRX-1010-E1 gateways, the switch of thesecond level D-Link DES-1024, 3CX Phone software for PC.

8.2.2 Assign IP address and subnet mask.8.2.3 Connect network cable to console port of VRX-1010-E1 gateway via

SSH-client.8.2.4 Connect network cable to console port of VRX-1010-E1 gateway via

SSH-client.8.2.5 Connect multimedia consoles to PC.8.2.6 Run 3CX Phone on PC and configure it to connect to gateway.8.2.6 Make test calls between PCs via the signaling protocol a connection

link, based on the PRI.8.2.7 Create a report.

8.3 Guidelines

8.3.1 Build the circuit as shown in figure 8.1

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Figure 8.1 –Network diagram

8.3.2 Configure according to exercise 2 VRX-1010-E1 for any gateway.8.3.3 Configure according to exercise 3 VRX-1010-E1 for any gateway.8.3.4 Add to the file /etc/asterisk/extensions.conf next linesexten => _8X.,1,Dial(SIP/<X>/{EXTEN:1})same =>n,Dial(DAHDI/g1/{EXTEN:1})same =>n,HangUp()exten => _7X.,1,Dial(DAHDI/g1/{EXTEN:1})same =>n,Dial(SIP/<X>/{EXTEN:1})same =>n,HangUp()Where <X> - remote gateway name according to an option.8.3.5 In console, enter the following command service asterisk restart.8.3.6 Make test calls between PCs on different gateways via prefixes of 7, 8,

9, 0. Disconnect E1 channels, Ethernet alternatively. 8.3.7 Create a report.

Table 8.1 –Initial data№ SIP-address № SIP-address № SIP-address1 1000 GW1 8 1007 GW2 15 1014 GW12 1001 GW2 9 1008 GW1 16 1015 GW23 1002 GW1 10 1009 GW2 17 1016 GW14 1003 GW2 11 1010 GW1 18 1017 GW25 1004 GW1 12 1011 GW2 19 1018 GW16 1005 GW2 13 1012 GW1 20 1019 GW27 1006 GW1 14 1013 GW2

8.4 Quiz

8.4.1 When to use restart command? 8.4.2 Describe features of linux? 8.4.3 What management interfaces do you know ?8.4.4 What does console mean?8.4.5 Network addresses and its purpose?

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Abbreviations

PBX – private branch exchangePC – personal computerBRI (Basic Rate Interface) – basic accessЕ1/Т1 – bit stream standards of European / AmericanIAX (Inter-Asterisk eXchange protocol) – xchange protocol of VoIP data

between IP- PBX Asterisk IP – Internet protocolISDN PRI – digital network with integration of services the primary interfaceIVR – Interactive Voice ResponseSFTP (SSH File Transfer Protocol) – SSH data transmission protocolSSH – «Secure Shell»SIP – Session Initiation ProtocolQoS – Quality of serviceXML – Extensible Markup LanguageVoIP – Voice over IPWinSCP – Graphical SFTP client

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References

1 Баскаков И.В., Пролетарский А.В., Мельников С.А., Федотов Р.А. IP-телефония в компьютерных сетях: Учебное пособие – М.: Интернет-Университет Информационных Технологий; БИНОМ. Лаборатория знаний,2008. - 184 с.

2 Гольдштейн Б.С., Зарубин А.А., Саморезов В.В. Протокол SIP – СПб.:BHV-Санкт-Петербург, 2005.

3 Яновский Г. Г. Качество обслуживания в сетях IP // Вестник связи –Алматы, 2008. – № 1. – С1-15.

4 Романчева Н.И. Современные Интернет-технологии: Учебное пособие.- М.: МГТУ ГА, 2007. – 104 с.

5 Чежимбаева К.С., Мирзакулова Ш.А. Основы IP телефонии:Конспект лекций. – А.:АУЭС, 2014. – 50 с.

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AUTHORS: Chezhimbayeva K., Mirzakulova S., Bublik V. Basics of IP-telephony. Guidelines for laboratory works for students of specialty 5В071900 –Radio engineering, electronics and telecommunications. - Almaty: AUPET, 2016. - 32p.

Laboratory works on subject "Basics of IP - telephony" are presented. Labworks on SIP protocol, ISDN interfaces, Asterisk and FreeSWITCH software areincluded.

Figures - 23, tables - 8, references- 5 sources.

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Reviewer: head of FL chair Kozlov V.

Published according to the plan of non-profit joint-stock company "Almatyuniversity of power engineering and telecommunications" for 2015 year

© NPJSC "Almaty university of power engineering andtelecommunications", 2016 y.

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