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Service Improvements for a VoIP Provider

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1. ssssssssssssseeeeenernnnrrrrrrrrrrrrrnnnrrrnnnnnn rn n nn nr rr rn rn rn rr nn nn nn nn nn 35 VII List of Tables Table 1 Detail explanations of SIP Messages ssmmussssssssseseerressserersrerererrrrrrrrrrrrrrrrrrrrrrrrrrrrrr rss rss rss rss rr rn VIII List of Abbreviations AGI API ARA AWT CDR CTM DI DoD DTMF EJB GPL GPRS GUI Java EE JBoss JDBC JDK JNDI JPA JPQL JSPs JVM J2SE IP IP PBX ISP NAT ODBC PBX PLMN PSTN RTP SDP SIP SMS SOA TLS UAC VoIP VPN Asterisk Gateway Interface Application Programming Interface Asterisk Realtime Architecture Abstract Window Toolkit Call Detail Record Companhia de Telecomunica es de Macau S A R L Dependency Injection Dial over Data Dual Tone Multi Frequency Enterprise JavaBeans GNU General Public License General Packet Radio Service Graphical User Interface Java Platform Enterprise Edition JBoss Application Server Java Database Connectivity Java Development Kit Java Naming and Directory Interface Java Persistence APIs Java Persistence Query Language Java Server Pages Java virtual machine Java Platform Standard Edition Internet Protocol Internet Protocol Private Branch eXchange Internet Service Provider Network Address Translation Open Database Connectivity Private Branch eXchange Public Land Mobile Network Public Switched Telephone Network Real time Transport Protocol Session Description Protocol Session Initiation Protocol Sho
2. 3 3 Three components of the DoD solution 3 3 1 Symbian client Java is platform independent language thus software using J2ME can run on almost all mobile devices equipped with Java However J2ME expect on the Android platform cannot detect and control the operating system in a comparable manner as a Symbian application can Thus J2ME cannot detect an outgoing call send DTMF and terminate a call Using J2ME is possible to implement a client for call back and call through but the user cannot use their traditional method of placing calls but instead explicitly launch Java client first then enter the necessary numbers and finally choose which method call back or call through to use This requirement of starting a Java application is viewed as an impediment to the routine use of the DoD solution hence the need for a Symban client The Symbian client is software running on top of a Symbian operating system platform This software is an optional component in the DoD solution but it helps users to place calls The Symbian client can be used to place call back calls and or call through calls This software is written in Symbian C which is a system level programming language of the Symbian platform Using this client user can place calls in the traditional way 1 e by simply entering a number and pressing the green dial call button The Symbian client detects this event and can process the call using the calling method selected by the use
3. Kevin Mukhar and Chris Zelenak with James L Weaver and Jim Crume Beginning Java EE 5 From Novice to Professional Apress October 2005 ISBN10 1 59059 470 3 ISBN13 978 1 59059 470 4 24 Wu Xiao SIP on an Overlay Network Master thesis Royal Institute of Technology KTH 54 School of Information and Communication Technology Stockholm Sweden expected August 2009 25 Sun www http java sun com products ejb Last access on 2009 5 25 26 Raghu R Kodali and Jonathan Wetherbee with Peter Zadrozny Beginning EJB 3 Application Development From Novice to Professional Apress September 2006 ISBN10 1 59059 671 4 ISBN13 978 1 59059 671 5 27 Debu Panda and Reza Rahman and Derek Lane EJB 3 in Action Manning Publications Co April 2007 ISBN 1 933988 34 7 28 JBoss Application Server Official Documentation Page JBoss Application Server 5 Installation and Getting Started Guide www http www jboss org file access default members jbossas freezone docs Installation And Getting Started Guide 5 html Book Preface html Last access on 2009 5 25 29 Companhia de Telecomunica es de Macau S A R L CTM www http www ctm net Last access on 2009 08 02 30 CDR wiki www http www voip info org wiki view Asterisk cdr csv Last access on 2009 08 02 31 Asterisk config extensions conf www htto www voip info org tiki index php page Asterisk 20config 20extensions conf Last access on 2009 08 02 3
4. A they can call this number which was assigned belongs to user A Subsequently user A s local extension will receive this incoming call If user A enables his her followme feature which will be explained in next paragraph then the defined followme number will receive this call 3 1 3 3 Followme call Followme is a feature which enables callers to reach you anywhere Assume that user A has a local extension in this case a SIP user agent a mobile phone 0707100100 and a number 0757008811 assigned by the IP PBX Using the later number this user can be reached from the PSTN or PLMNs via the IP PBX Situation 1 User A has enabled his her followme feature set to his her mobile phone 0707100100 When someone calls 0757008811 from the PSTN or PLMNs both his her local extension and this followme number will ring together If he she is in his her office he she can pick up his her local extension If he she is outside office for example when traveling he she can answer his her mobile phone No matter which phone he she picks up both will stop ringing Situation 2 If user B places a call back call or call through call to reach user A s local extension and user A has enabled the followme feature then both user A s local extension and followme number will begin ringing Just as in situation 1 he she can pick up whichever phone he she wants to use Followme feature is designed for someone who often needs to travel or has multiple
5. access on 2009 5 25 9 SJphone www http www sjlabs com Last access on 2009 5 25 10 Nonoh www http www nonoh net Last access on 2009 08 02 11 VBuzzer www http www vbuzzer com Last access on 2009 08 02 12 VoIP Stunt www http www voipstunt com Last access on 2009 08 02 13 Fring www http www fring com Last access on 2009 08 02 14 Asterisk www http www asterisk org Last access on 2009 08 02 15 Asterisk wikipedia www http en wikipedia org wiki Asterisk PBX Last access on 2009 5 25 16 J Rosenberg H Schulzrinne G Camarillo A Johnston J Peterson R Sparks M Handley E Schooler SIP Session Initiation Protocol IETF RFC 3261 June 2002 17 Henry Sinnreich and Alan B Johnston Internet Communications Using SIP Delivering VoIP and Multimedia Services with Session Initiation Protocol an Edition Wiley August 2006 ISBN 0 471 77657 2 18 Luan Dang Cullen Jennings and David Kelly July 2002 Practical VoIP using VOCAL O Reilly ISBN 10 0 596 00078 2 ISBN 13 9780596000783 19 voip info www http www voip info org wiki view Asterisk sip t Last access on 2009 5 25 20 Jim Van Meggelen Leif Madsen Jared Smith September 2005 Asterisk The Future of Telephony O Reily ISBN 10 0 596 00962 3 ISBN 13 9780596009625 21 voip info Asterisk CDR Fields www http Last access on 2009 06 18 22 Sun www http java sun com javaee Last access on 2009 5 25 23
6. both the DoD server and the IP PBX DoD server uses made by JavaEE We use JBoss as the JavaEE container because JBoss is open source software and free to use The DoD server provides a web GUI for placing call back calls Additionally the DoD server can place call back calls based upon processing HTTP or HTTPS requests sent by clients The DoD server also can change some of Asterisk s settings thus we do not need additional GUI for configuring Asterisk Using one GUI for both placing calls and configuring Asterisk 1s an improvement that will be explained in more detail in next cheaper In addition to calling and management features the DoD server also provides some extra features such as a contact book for simplified dialing and a GUI for showing CDR records as a call history The following figures are the some of screenshots of DoD server 28 Current user optical Role Administrator Call Address book Address book manager User setting Statistics Make a call Linel Line2 Figure 13 The web GUI for call back Current user optical Role Administrator Logout Call Address book Address book manager User setting Statistics Contacts call contact call contact call contact Figure 14 The web GUI for contact book 29 Current user opticall Role Administrator Users Statistics Settings PEX Number Username Role Jorgen e a suntao manager zhangh user Figure 15 The web GUI for administrator
7. call histo T 0 036 1 218 5 203 0 932 1 163 ogn T 0 178 2 955 10 937 2 367 2 16 togaut_T 0 007 0 128 2153 0 012 0 32 zettings_T 0 061 2 675 FART 2 596 1 593 statistics T 0 053 2 687 8 545 2343 2 06 Figure 26 Average response time for each page in second 5 2 5 Problems faced in Tests During the testing we faced some problems When there are more than 200 users the DoD server throws many java net SocketException Too many open files exceptions This is because Linux has a limit for the maximum number of open files By default Linux allows 1024 open files at the same times We can use command ulimit n to check this limit and use ulimit n num to set it We should examine what would be a better setting for this limit There is also a limitation in JBoss as the configuration file server xml defines how many connections JBoss can handle The default value is 200 thus we should increase this value to enable testing with more than 200 users As noted in the previous section the response times with more than 150 160 users are unacceptable thus neither of these parameters limited our testing However for future tests with a more powerful server these limits must both be increased 46 5 3 Test for MySQL The database has an important role in DoD solution as it is used by both the DoD server and Asterisk An advantage of this approach is that users can change their own settings by simply changing their e
8. can provide cheap calls for all users which are connected by Internet The users can initialize and tear down a session using a VoIP user agent but how can they place a VoIP call from a mobile device or other devices without a VoIP user agent Users want to place cheap VoIP call everywhere VoIP providers want to provide flexible solution to attract and keep users So they both want to the users to be able to place cheap VoIP call everywhere Although VoIP user agent are available for many devices as a software running on a computer a hardware VoIP phone and even in some mobile devices However there are some practical problems with placing a VoIP call from everywhere The first problem is that not every device can have a VoIP user agent But if you do not have a VoIP user agent on your device then it would seem to be difficult to place a VoIP call The second problem is that you have to connect to a network probably Internet to signal that you want to place a call Thus at a minimum your device has to support connecting to an appropriate network If your device is connecting to a mobile network you can send signaling to set up a VoIP call through General Packet Radio Service GPRS However the bandwidth and delay of the GPRS networks of some mobile operators is not suitable for the transfer of encoded voice data additionally some mobile operators charge high fees for using GPRS All of these problems make placing VoIP calls via a mobile device di
9. dynamic ARA allows you to change the configuration in the database without requiring the application to reload all of its configuration information and re configure itself This is very useful especially in a SIP setting For example it is very likely that you will need to add SIP users over time By using a dynamic ARA you can add these users without reloading the entire system configuration which would mean that service would be interrupted during this period of time However not all Asterisk configurations support dynamic ARA 2 2 1 3 Web GUI for configuration There are many web based GUIs for configuring and managing Asterisk Using a GUI is very convenient for a user to manage Asterisk as the user does not need to manually change configuration files or manually reload Asterisk In most cases the user can simply click to configure items then click to apply the changes You can even use Open Database Connectivity ODBC to store configurations in databases which support ODBC 2 2 2 Call Detail Records CDRs Call Detail Records store information about calls which were processed by Asterisk CDRs can be stored in databases Subsequently an operator of the IP PBX can use these CDRs to generate billing information There are many fields in a CDR for a complete list see 21 the following are the most important fields for billing accountcode an unique string up to 20 characters identifies a subscriber who should pay for this
10. rdvara VoIP telefon och ven 1 vissa mobila enheter Men det finns vissa praktiska problem med att ringa ett VoIP samtal fr n alla platser Det f rsta problemet r att inte varje enhet kan ha en VoIP user agent Det andra problemet r att den m ste ansluta till ett n tverk troligen Internet f r att signalera att den vill ringa ett samtal Om din enhet ansluter till ett mobiln t kan du skicka signalerar att uppr tta ett VoIP samtal via General Packet Radio Service GPRS Dock r bandbredden och f rdr jningen 1 GPRS n t 1 vissa operat rers n t inte l mpliga f r verf ring av tal dessutom tar vissa mobiloperat rer ut h ga avgifter f r att anv nda GPRS Alla dessa problem g r det sv rt att hantera VoIP samtal via en mobil enhet Men om din mobila enhet har en VoIP user agent och du har l mplig n tanslutning s kan du enkelt anv nda VoIP fr n din mobiltelefon F r att erbjuda en flexibel VoIP l sning verallt ven p enheter som inte kan ha en VoIP user agent har OptiCall AB utformad Dial over Data DoD Genom att anv nda denna l sning kan du initiera ett VoIP samtal fran din mobiltelefon eller fast telefon utan att kr va att den enhet som du anv nder har en VoIP user agent Denna l sning inkluderar ocks en central Internet Protocol Private Branch Exchange IP PBX som kan koppla samtal till och fr n Session Initiation Protocol SIP telefoner B de privatpersoner och f retag kan anv nda denna l sn
11. s side there are two main ways to make a VoIP call via the IP PBX Call through and Call back I will explain these two methods in section 3 1 Use of a SIP user agent is also supported The SIP user agent can connect to the IP PBX which acts as a SIP proxy in order for the user to place a VoIP call to callee Using SIP and RTP many manufacturers have implemented their own VoIP hardware as well as software Asterisk 1s a software implementation of a telephone PBX originally created in 1999 by Mark Spencer of Digium Like any PBX it allows attached caller to make calls to one another and to connect to other telephony services including PSTN 15 Additionally it enables calls to and from VoIP service providers Since Asterisk is an open source software PBX many VoIP providers using the SIP protocol use this software PBX Asterisk is free to use hence we do not need to pay any fee for using it in commercial products Additionally Asterisk supports a variety of features and supports a number of devices that are of interested to us Therefore Opticall AB chose Asterisk as its software PBX This DoD solution have been developed by several persons The complete DoD solution is up and running now but there are some issues and some parts that need to be improved For example the call detail record CDR information is in chaos there are two different web Graphical User Interfaces GUIs for controlling this solution and caller
12. the dialplan in the database The table structure is fixed and includes all the necessary information for the dialplan Each record is an extension specifying a priority and command In order to identify which extensions belong to which context each record also has a column to save context information see Figure 19 Although using ARA for dianplan has many advantages such as enabling changesto the dialplan without reloading easy access for external application and so on but there is also a limitation Figure 19 shows part of a dialplan as stored in the database while Example 10 shows part of a dial plan stored in a flat file Example 10 each row is an extension containing an extension number priority and a command Unlike the case of a flat file where the order in the file could be used to specify an implicit priority and order in the cases of storing the dialplan in a database we have to explicitly specify the priority ordering of the records that concern a given extension Despite this small amount of extra work we believe that ARA offers many advantages over using a flat file 34 id context rn m m 1 mm su gomm m m D hi OS d UL Cl TO CALE IHU SCA Clay SIP EXTEN mx 52 disatocal x 3 hangup 55 disa outgoing x 1 Noop disa outgoing call to number S EXTEN 62 disa outgoing x 2 Set CDRiuserfield bill 60 disa outgoing x 3 dial SIP sting S EXTEN 62 disa outgoing xX 4 hangup 37 followme xX 1 N
13. traditional method of dialing to place calls The IP PBX validates the incoming number of user A and answers the incoming call At this point the Symbian client sends Dual Tone Multi Frequency DTMF tones to the IP PBX to tell it the destination number user B s number see Figure 4 step 2 Finally the IP PBX places a call to user B After B answers this call the IP PBX bridges these 17 two calls see Figure 4 step 3 DoD Server Step l DN Step 3 IP PBX calls A calls IP PBY s us B and bridges the number ii Step 2 A Sends B s number using DTMF O gt Figure 4 Call through procedure The Symbian client is not essential to using call through Users can manually dial the access number of the IP PBX After the IP PBX validates the user and answer the incoming call the user can manually dial the number of the destination Thus the user can use any brand of mobile phone including those which are too old to install new software This same method can be used 1f the user wants to place calls from fixed phones However the weak point of placing a call through calls without a Symbian client is that users have to remember two numbers the access number of the IP PBX and the number of destination when dialing and they must explicitly dial the access number of the IP PBX first and wait until this call is answered before dialing the actual number of the party that they wish to reach We can see that the call th
14. when users enable their followme feature or following another administrative change If we have to rewrite the configuration files and reload settings when any settings have been modified this is both a lot of work and inefficient Therefore we decided to use the Asterisk Realtime Architecture ARA to solve this problem As mentioned at subsection 2 2 1 2 using ARA the configuration can be saved in a database ARA also supports dynamic updates of the configuration without requiring reloading For example if you want to change a SIP account you only need to change it in database Another advantage of this approach is that application can change or manage the Asterisk configuration s This can be quite simple but very power since the application can change the behaviors of Asterisk simply by changing entries in the database In DoD solution we save SIP and diaplan settings in a database because these two kinds of setting may be changed frequently In order to save these setting in a database we need to create database tables for each one The structure of each table is constant and includes all the information necessary for each kind of setting After we creating two tables in the database we can add records to these tables For SIP each record is a SIP account When we save the SIP settings in a flat file we entries of the form key value to define a SIP count see Example 9 Each key represents an attribute of this acco
15. work places Users can enable this feature and set their followme number as their mobile number when the user is traveling Similarly the user can set their followme number as their home number when 21 they are working at home in order to receive incoming calls on their fixed telephone 3 2 Caller ID In order to show the correct caller IDs to different devices we change the caller IDs to the user s assigned extension before the IP PBX places the call to user B If user A uses the DoD solution to place a call to a mobile or fixed phone the number that will be visible to the callee is the number that user A was assigned by the IP PBX There many different situations in which the DoD solution should display a different caller ID than the actual caller ID of the user The following subsections and figures will explain when and how we display these different caller IDs In all of these examples we assume user A has mobile number 0707100100 and a number assigned by IP PBX which is 0757008811 The access number of the IP PBX is 0757008810 User A will use this access number to place a call through call User A s local extension is 8811 We will also assume that user A has set his her followme number to 0707200200 3 2 1 Caller ID when using Call through When user A places a call using call through to a mobile number the caller ID CID presented at the callee s mobile should be the number which is assigned by IP PBX See figure 6 As a r
16. 100 queries that could be sent at the same time Comparing the number of requests sent to the 2800 requests sent by the DoD server one might think that these additional database requests would not be a problem However this is not true since the load testing of queries to the database by the DoD server were spread over more than six seconds while the maximum load of the SIP trunk could all come within less than one second Therefore if the calls come in during a short period of time there may be a performance limit caused by the SIP trunk There is also a problem with the assumptions about the SIP trunk and our performance measurements Each call coming into the DoD server could cause one or more calls to go out For example if the callee has a fixed phone and a cellular phone and is using the FollowMe functionality then two outgoing calls have to be made Depending upon whether we are using early media or not this may require 0 1 or 2 calls capacity from the SIP trunk If the caller is coming in via the PSTN or PLMN and placing a call that must also go out via the PSTN or PLMN and if these calls are carried via the SIP trunk then because of call bridging we need to allocate SIP trunk resources for the duration of these two calls We have not properly tested the performance of the Asterisk server or SIP trunk hence these should be tested more thoughoutly in future tests Additionally we might be able to leverage the testing that o
17. 2 0703008177 2009 07 21 10 46 05 20 seconds Jopticall 0703008177 0707539452 2009 07 21 10 47 56 18 seconds Isuntao 8813 fes12 2009 07 21 10 55 55 0 seconds tand 0017 loo12 hann nT 1 TPE manman ede Figure 20 Web GUI showing CDR for administrator As mentioned section 4 2 we support different caller IDs at different devices simultaneously by using a local channel The caller IDs can be correctly manipulated using this local channel but there are many extra CDR records generated in the database for each call Because we use a local channel an extra CDR record is created for each of these local channels Unfortunately users are confused by these useless extra CDR records especially when they see that many call happened 37 at almost the same time These unnecessary CDRs make it hard for the user to find the useful information In order to solve this problem we use the command NoCDR for the local channel call in the dialplan As a result Asterisk will not save the CDR for this call 33 Alternatively we could simply filter the CDR entries that are to be presented and show only those that are significant 4 4 Optimizing code The DoD server was designed to process HTTP requests from users actually from the user s clients to place call back calls and also to perform management tasks concerning both users and the IP PBX 4 4 1 Using EJB 3 replacing EJB 2 As mentioned at s
18. 2 Asterisk local channels www http www voip info org wiki view Asterisk localt channels Last access on 2009 08 02 33 NoCDR www http www voip info org wiki view Asterisk cmd NoCDR Last access on 2009 08 02 34 Annotations Sun www http java sun com docs books tutorial java javaOO annotations html Last access on 2009 08 02 35 Java Basics Model 2 www http javias net blog mu blog 2009 04 16 java basics model 1 and model 2 Last access on 2009 08 17 36 FreePBX www http www freepbx org Last access on 2009 08 02 37 Smoke testing www http en wikipedia org wiki Smoke_ testin Last access on 2009 08 18 38 Load Runner HP www 126 17 5E8 4000 100 _ Last access on 2009 08 18 39 LoadRunner to Performance Center Migration An Objective Analysis of Migration Options White paper HP 4AA1 4912ENW May 2008 40 JMeter Apache www http jakarta apache org jmeter index html Last access on 2009 08 19 41 User Manual JMeter Apache www http jakarta apache org jmeter usermanual build db test plan html Last access on 2009 08 19 42 Asterisk V1 4 11 Performance TransNexus www http www transnexus com White 20Papers asterisk_V1 4 11 performance htm 55 Last access on 2009 08 28 43 Asterisk Background performance test KOLMI soft www http wiki kolmisoft com index php Asterisk Background performance test Last access on 2009 08 28 44 Oracle WebLogic www http www oracle co
19. 2 5 Providing machine readable call history ssssmm0sssrsssrrrrsreresrrrerrrrrrerrrrrrrrrrrr sr rrrsr nr nns rn ren 52 FRG aro E A rost asken Ness NOAs dad seasons ARS ee 54 List of Figures Figure 1 Simple message flow in setting up and tearing down a SIP session between A and B 5 Figure 2 Multiple call detail records when placing CallS sssssrerrsrrrrrrrrrrrrrrrrrrrrrrrererrrrrrrrrrrrrrrrrrrn 6 Figure 3 Web application structure in JBOSS cccceceesssssecceeeeeeeeeeeeeseeseeeesnneaaeeeeeeeeeeeeeeeeeeeeeeegs 16 Figure 4 Call through procedure sssssssssssrssrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrs rss rss rss rr ere rese ner rr rr rr enes sr rer nr rer ties 18 Fisure 5 Call Dack PrOCe Ule sesacescsinadnsasasabenedsdnnndeiedeascadsduooaeosboasdedadaasduaedadonedsdacubedea ndteeasdeasenandedaet 20 Figure 6 Showing the user s IP PBX assigned caller ID when using call through 22 Figure 7 Showing different caller IDs when using call through between two users connected to the same DoD Syste 1 eo eee n eee n ee eee ene ee ee ee eee 23 Figure 8 Showing caller ID when using call back mssmnsssssssssssssssssrrrrrrrrrrrrreressssrrrrrrrrrrrrrrrrrrrrrrrrnnnna 23 Figure 9 Showing caller ID when using call back mssmnsssssssssssrsssssrrrrrrrrrrrrreeessssrrrrrrrrrrrrrrrrrrrrrrrrn nn na 24 Figure 10 Showing caller ID when using SIP User agZeDt ssssseessesserrrrrrrrrrrrrrreressrrrrnrrrrrrrrrrrrrrrnrr rna 24 Fig
20. 811 CID 0757008811 Local extension 8811 CID 0757008811 Figure 8 Showing caller ID when using call back If user A places a call back call to user B s local extension this assumes that both users are in the same DoD system The local extension of user B will receive a call with the caller ID of user A s local extension If user B has enabled their followme feature the followme phone will receive a call indicating as the caller ID user A s IP PBX number See figure 9 23 o e Mobile 0707100100 OCs exe aston PBX number 0757008811 p CID 8811 Local extension 8811 CID 0757008811 NN Ly gt C gt Followme number CID 0757008811 Figure 9 Showing caller ID when using call back As can be seen in both the call through and call back case the called user sees a consistent call ID 1 e it is always the number that they can use to call the caller back Additionally in the case of calls sent via public telephony networks the caller IDs are all valid caller IDs for the party making the call to use However there is a clear difference between this behavior and the behavior that a caller and callee who have the same mobile operator and are members of a closed calling group would see in this case the caller and callee would both use and see only a local extension number 3 2 3 Caller ID when using SIP call Users also can use a SIP user agent to directly place calls If user A places a call using their
21. Asterisk The Symbian client is used to help a user to place a call via DoD from a mobile phone The DoD server is used to process requests from both users using computer and Symbian clients and present web interfaces to users The IP PBX is used to route outgoing calls and receive incoming calls The IP PBX is assumed to be connected to an upstream VoIP provider who is connected to one or more PLMNs and PSTN so the calls from to IP PBX will go through this VoIP provider This chapter will explain these three components in detail 3 1 Two methods to place VoIP calls From the user s point of view the user simply wants to place and call and do so at low cost The user does not care about how this is accomplished provided it does not take significant extra effort DoD solution providers two primary methods to place such calls call through and call back 3 1 1 Call through We assume that a user wants to use a mobile phone of Symbian platform to place a call After the user dials the destination number and presses the green Call button the Symbian client will detect this action The Symbian client stops the normal processing of this action at the operation system level Instead of dialing this number and placing a call via the PLMN the Symbian client causes the mobile phone to dial the access number of our IP PBX see Figure 4 step 1 The Symbian client makes step is transparent for users so users can continue using their
22. BC requests in a file and presenting a visual model of the data 41 5 3 3 Test result Figure 27 shows the results of each simulated user making 4 queries with an average of xx seconds between query requests 47 o 40 Ln oo wil aii S100 i el E Ss Si i s E Wi i fa ease eee ian 1 6 216 second query response time response time average Figure 27 Response time to query the database This testing includes 2800 queries based upon 200 user X 7 pages X 2 queries of database The average time of database query takes 58 96lms The test duration was 6 216 seconds this was enough time to complete all the queries for all 200 users Note that despite the queries being spread over more than six seconds the maximum query response time is more than 1 450 seconds which leaves very little other time for the system to perform other operations while maintaining a response time as seen by the user of less than 2 seconds 5 3 4 Test Analyze There were no errors reported during this test All the queries got their correct responses While the average time to perform a database query was 58 961 milliseconds which is shorter than the response time of pages the maximum query times are not so encouraging Therefore further testing 1s needed to understand who to reduce this maximum query delay by an order of magnitude or m
23. ID is not displayed properly One of the goals of this thesis project was to determine which parts of the existing solution needed to be improved how they should be improved and to implement and evaluate these improvements Following this initial chapter chapter 2 provides background information about the three main technologies that are relevant to this thesis SIP Asterisk and Java EE Chapter 3 describes the current DoD solution Chapter 4 examines the areas that need improvement and presents the improvements that have been made in each of these areas Chapter 5 evaluates these improvements Chapter 6 presents some conclusions about this work and suggests some future work Chapter 2 Background This chapter will explain some of the technologies which are used in the DoD solution This chapter begins with an explanation of these technologies and some equivalent technologies This chapter also explains which parts of these technologies will be used and how they are used in the DoD solution 2 1 SIP protocol SIP is an application layer control protocol that can establish modify and terminate multimedia sessions conferences such as Internet telephony calls 16 The following sections will elaborate some of the relevant details of SIP For a general introduction to SIP the reader is encouraged to read Sinnreich and Johnston s book Internet Communications Using SIP Delivering VoIP and Multimedia Services with Session Initiation
24. ING Call ssmssmsssssrerssrrrrrrrrrerrrrrerrrrrrrrrrrrrrrrrrrrrrrrrrrrr rr sn 25 3 3 Three components of the DOD solution s sssesssseressrrersrrrrrrrrerrrrerrrrrrrrrrrrrrrrrrr sees sr rens rr rr n ren 26 Meo tel Few 1 04 6 SEC T gee eee eee rr 26 ISA WP BA MA ST RISK ee otannataescseaatencnaaidannmuaaansmaenesaniaceneecautee saacvauaeens 26 ig oO DES ET CU er cress cate ep atecnnng E E EE 28 hare A WFO VSI GIA S a E oaasnannencuerausaieautenenreseceann 33 4 1 Saving Asterisk configurations IN database esssseeessserrsrrrerrrrersrreresrererrrerrrrrrrrr rer rr rrrrrr rr en 33 MM Mer D e od rr ee E RSS enuade nesansn aeseetneseaenaeeesiente ened 35 eS OVINE CDR TE GOT CS oinei nena sepunteanetaosaaiteaaaai tac utec ance ueaeoosuinestreniuanesenveanaaad 36 A OCI MIMS COGS Sieen EE EE S r 38 44 1 Using ENB 3 replacing EJB 2 as ssutsecossctaesnnaneserdesiie cesautwas vanttcanaantnssenonieanenensbauncassuateeses 38 4 4 2 Using Model view controller mode 2 osmmssessserersrrrrsrrrerrrrerrrrrrrrrrrerr sees rr rrrr rr ere rr rn rr 40 4 3 Using one GUI for both calling and configuring ssmmeesserssrrrersrrrrrrrrrrrrrrrrrrrrrrr nere rrrrr rr rn 41 Chapter S EV AN at OW esscetceacnecnctciacuancesanscesmnntacnpintneesaacnoasenennes oauaioedesuntaouerssdoraseeeaconrerewantnenatenies 42 DL Test EAV POMC NU ssai er aa a NEA EEE E RE AE ARER S 42 5 2 Tests for JBoss Application Server DOD SErVEr ssssserseossrrrrrrorrrerrrrrrrerrrrrrerrrrrrrra 43 DE
25. JB entities are responsible for manipulating a database Instead of using complex Java Database Connectivity JDBC code you simply map your database tables to entities These entities are actually Java objects Using these objects it is possible to perform CRUD creating reading updating and deleting operations on a database by manipulating the Java objects In the latest version of EJB entities no longer belong in the EJB specification Instead the Java Persistence APIs JPA handles the entities on behalf of the programmer JPA provides a SQL like language called Java Persistence Query Language JPQL for performing CRUD operations 2 3 1 2 Advantages of EJB version 3 There are a number of advantages to using EJB version 3 Specifically we will take advantage of XML and annotations and metadata based Dependency Injection Each of these will be described in more detail below Annotation and XML One of the hallmarks of the EJB component model is the ability for developers to specify the behavior of both enterprise beans and entities declaratively as opposed to 12 programmatically using their choice of Java Development Kit JDK 5 0 annotations and or XML descriptors 26 In EJB 3 a programmer can use both XML and annotation to configure an EJB or an Entity Using an annotation to configure an EJB is very easy and clear because the annotation is located in your Java code However some programmers still like using XML because
26. JSP Model 2 is the new approch to overcome the shortcomings of Model 1 MVC Model View Controller which is the fundmental of Model 2 is the concept which has been used in the GUI development such as Applet There are already many web framework with MVC in place these days JSP is the entry point in Model 1 whereas Controller Servlet 1s the entry point in Model 2 35 Also some pages are not well organized because some logics which should be in the business layer are also in the presentation layer For example one JSP page includes almost 1000 lines of code In order to make the code easy be to maintained we changed the structure to model view controller model 2 architecture This model has three layers with each layer having its own responsibility The difference between a model and model 2 architecture is that the request from users is not sent directly to the presentation layer Instead there is a controller that receives request from the user and pass this request to the business layer or directly to presentation layer By using model view controller model 2 architecture we can reduce the amount of logic code in the presentation layer because the controller can process the requests and get information needed for the presentation layer Therefore a model view controller model 2 architecture is better than a model view controller model 1 architecture especially for web applications 40 4 3 Using one GUI for both calling and configuri
27. LTO I e E E 43 Did 2 TTE COO I o E E EE esaasieecaciateruesreneneartiee 44 De TEST TE SUN E E O nns ine 44 5 2 4 Analysis of the test S reSUlt S snnssssessssersesrrrererererrrrersrerererrrrrrrrrrrrrrrrr rens sr rr rr r rss rr rss rn sn 45 5 2 5 Problems faced in TEStS ssssossereesrrrerrrererrereerrererrrrerrrrrerrr reser sees sr rer rss rer s RR RR RKS RR rss renen 46 SoTa oNV S CE E E 47 Dy TEST L T a EES 47 SE PTE COO E E nns soner EAE E E E T sc Os ANOS ANS MENN 47 SS TES o l a E rk NANA eb 47 AEE AE E A 48 5 4 Evaluation of Asterisk and the SIP trunk ssosososessrsrsrrrrrrsrsrrererrrersrrrsr rr rr rr rr rr rr rr ARR 49 So 8 11 6 Oh a gj ee T eee ee eee REN ARENA SUNNE 49 Chapter 6 Conclusions and Future WOTK ssserssrererrrrrersrrrrrrrrrrrrrrrerrrrrrrrrrrrrrrrrrr ers rens rr rr rna 50 PECON UE ON teeters nase tte eeeiaonc A T A E aia 50 AFOS WO Kene E Nesrin 50 6 2 1 Using a framework in the presentation layCr s sssesserrsrrrrrrrrrrrrrrrrrrrrrerrrrrrrrrrrrrrrrr ren 51 6 2 2 Multiple access numbers for the IP PBX sssccccsssececesseececeesececeeeseceseeeceeeeeceesenes 51 6 2 3 Other ways to place call back calls SMS and VOICE ssssrrersrrerrsrrrrrsrrrrsrrrrerrrrrrrrn 51 62A ESMS Call Da CK e N 52 62 Z VOICE CIl DICK roria a tubule de dule tub gsteranderaus indre rskr 52 6 2 4 Increasing security for call through s sssssssersssrrrrrrrrerrrrrerrrrrrerrrrerrrrrrrrrrrr sr rr brer rr rr rn rr 52 6
28. Protocol 17 2 1 1 What SIP can do We normally use the SIP protocol for setting up VoIP calls When two parties want to talk with each other the SIP user agents send SIP messages to negotiate and establish a session However SIP does not carry the media content After a session has been established the SIP user agents use RTP to communicate based upon the information that was passed using the Session Description Protocol SDP in SIP message bodies Finally SIP terminates the session when one of the parties hangs up 2 1 2 SIP Components In SIP there are several types of components They are classified into two categories SIP user agents and SIP servers e SIP user agent A SIP user agent could be implemented by software or hardware There are many software SIP user agents such as X lite SJPhone and miniSIP There are also many hardware SIP user agents often called IP phones By using a SIP user agent you can establish a session to communicate with another party or parties The SIP user agent helps you to create a session using the SIP protocol then transfers and receives media to from the other party parties using RTP SIP servers SIP user agents can work without a SIP server as they can communicate with each other directly using their IP addresses But in some situations these SIP user agents cannot find each other as they do not know the other party s current IP address SIP servers help SIP user agents to
29. SIP user agent to someone s mobile phone then the callee s mobile will receive a call with a caller ID being the number assigned by the IP PBX to user A See figure 10 Someone s mobile CID 0757008811 SIP user agent Local extension 8811 Figure 10 Showing caller ID when using SIP user agent If user A places a call using their SIP user agent to user B s local extension the caller ID presented at user B s local extension should be the local extension of user A If user B has enabled their followme feature then the followme phone will receive a call showing as the caller ID the number assigned by the IP PBX to user A See figure 11 24 R AR 9 AL IP cal GS 0 Local extension SIP user agent 0 a Local extension 8811 A CID 8811 a Followme number CID 0757008811 Figure 11 Showing caller ID when using SIP user agent 3 2 4 Caller ID when receiving Incoming call When someone calls user A s IP PBX number the local extension of user A will receive a call showing as the caller ID the real number of the caller If user A has enabled their followme feature then the followme phone will receive a call also showing as the caller ID the read number of the caller See figure 12 Incoming call To 07570088 l Local extension 8811 CID 0707123123 Someone s mobile 0707123123 4 a Followme number CID 0707123123 Figure 12 Showing caller ID when receiv
30. SIP works in the DoD solution s sssssseressrrrersrrersrererrrrrrrrrrrerrrrrrrrrrrrrrrrrs rr rss reses rr rn n sn 6 IO TS 2 Go Gee ae ee E RE Ca ee ne ee oe ee ee 6 2 2 1 How to configure an Asterisk server snsssmm0ssserssrererersrsrrerererrererrerorerersr resor erern beror renas 7 2 22 Call Detail Records CDRS haseran inet dns 10 2 2 5 Asterisk Gateway INCI ACE sisemine 10 2 53 JAVA ER ETI ova vars season EAEE ET AEE aaa deer eins E EE deeandans E E ES 11 PA TEJP en E E ee eC ee een eee ere er ee 11 PINE BRN OSS ond cee caine fovaivos dione et dtencoug E 15 chanter o DoD SONU ON irene teen cs saeatene cena negesenessatgeeesiauesoiaesannsoaetecad nave cusarsasvatonedsenseeranesestare 17 3 1 Two methods to place VOIP CallS ccsesecsosssccsosssscnecssusecccussscucussscnenssessuesecuenesseneues 17 SLEP OUE iei E 17 MEAE D o E A E E E E E E T ee ee ee A 19 zL Other calme UMOOS ess susncd asi seccomswn tos onnnnatensseceaeecnscaenanesaeantecousensosetasnwnsesecssesanasonueet 20 BD CAMO p IB E A E san coeaesouetas essen AE A E T TT 22 3 2 1 Caller ID when using Call through ssssmmssssessssrrerrrrrersreresrrrrrsrrrrerrrrerrrrrrrs rer rs rens rr rr s resa 22 3 2 2 Caller ID when using Call DaCKk ssssssssssssrrersrrrerrrrersrererrrrrrrrrrrrrrrrrrrrrrrrr rer rr ers s rr rss rr sn 23 5 2 3 Caller ID when using SIP Call sssssssmsteneisesssuselonss nissvsnonssassserbonsninisssensntisont ssirbssksta 24 3 2 4 Caller ID when receiving INCOM
31. Service Improvements for a VoIP Provider Gp pa ZHANG LI KTH 3 VETENSKAP O98 OCH KONST A IOWA KTH Information and Communication Technology Master of Science Thesis Stockholm Sweden 2009 TRITA ICT EX 2009 104 KUNGLIGA TEKNISKA H GSKOLAN Final Report Royal Institute of Technology Version 1 3 Date 08 29 2009 Service Improvements for a VoIP Provider Master Thesis Final Report Zhang Li lizhang kth se 2009 08 28 Examiner Professor Gerald Q Maguire Jr Supervisor Jorgen Steijer Opticall AB Stockholm Kista Isafjordsgatan 22 Abstract This thesis project is on helping a Voice over Internet Protocol VoIP service provider by improving server side of Opticall AB s Dial over Data solution Nowadays VoIP is becoming more and more popular People use VoIP to call their family and friends every day It 1s cheap especially when users are abroad because that they do need to pay any roaming fee Many companies also like their employees to use VoIP not only because the cost of calling is cheap but using VoIP means that the company does not need a hardware Private Branch eXchange PBX while potentially offering all of the same types of services that such a PBX would have offered As a result the company can replace their hardware PBX with a powerful PC which has Private Branch eXchange PBX software to connect all the employees and their VoIP provider At the VoIP provider s side the provider
32. able to return to the dialplan after call completion 32 Thus when the IP PBX is going to place a call to more than one destination the IP PBX uses a local channel to place this call If the destinations of this call are two places this call will be rerouted to two different context and the caller ID can be suitably modified for each of these calls In order to use DoD server to change the dialplan without reloading we use ARA and save all the dialplans into a database For details about ARA see section 2 2 1 2 on page 9 3 Validate user and showing different CID using func_odbc In order to validate the user we use information stored in the database We use the func odbc in our dialplan The func odbc dialplan function allows you to create and use fairly simple dialplan functions that retrieve and use information from databases directly in the dialplan There are all kinds of ways in which this might be used such as managing users or allowing sharing of dynamic information within a clustered set of Asterisk machines When the IP PBX receives an incoming call we can use func_odbc to look up whether this caller ID is already in the database as that of a valid user Similarly if an outgoing call has multiple destinations we use func odbc to look up in the database the corrent caller ID to be used for each call 3 3 3 DoD server The DoD server provides web GUIs to allow a user to place call back calls and for the administrator to manage
33. all Source Call Destination Start Time ration O7OS00817 70707539452 2009 07 21 10 47 56 18 seconds 0703008177 8812 2009 07 21 10 58 47 5 seconds 0703008177 8812 2009 07 21 11 00 44 0 seconds 0757008811 0704648198 2009 07 21 11 55 31 0 seconds O7OSO08 1770707539452 2009 07 21 16 04 43 0 seconds 075700881110707539452 2009 07 21 16 07 19 8 seconds 0757008811 0707539452 2009 07 23 16 58 00 13 seconds O7O3008177 0707539452 2009 07 25 17 01 21 15 seconds HEART E 8811 88 14 2009 07 24 20 20 25 0 seconds D lt i LA lt i D D DO DO D lt i D lt i LA L d D d LA ro 2009 07 30 12 37 18 27 seconds Total time 1 minutes 26 seconds Figure 19 Web GUI showing CDR for normal users Overview lzhangk 32 seconds 0 seconds 13 minutes 34 seconds 0 seconds lopticall 27 seconds lo seconds 27 seconds lo seconds Detail Information Username Call Source Call Destination Start Time Duration chang 0707539452 0704648198 2009 07 21 10 24 03 0 seconds Isuntao J0757008813 8812 2009 07 21 10 28 49 0 seconds Isuntao 0757008813 0703008177 2009 07 21 10 28 49 23 seconds lsuntao 0757008813 0703008817 2009 07 21 10 29 39 7 seconds Isuntao J0757008813 8812 2009 07 21 10 29 38 0 seconds lsuntao 0757008813 0703008817 2009 07 21 10 38 32 seconds Isuntao 0757008813 8812 2009 07 21 10 38 33 0 seconds chang 881
34. call src the souce extension Because we support users to place calls from more than one device so users can know where this call is placed dst the destination extension By checking the destination extension we can compute the rate for this call billsec the call duration from answer to hang up time is measured in units of seconds 2 2 3 Asterisk Gateway Interface The Asterisk Gateway Interface AGI is a programming interface for using programming languages in a dialplan By using AGI you can do many tasks Most popular programming languages such as PHP Perl and Python can be used This feature is very useful as you can control Asterisk or communicate with other systems via a network or even a database in a dialplan Example 4 shows what an AGI script might look like agi request test py agi_channel Zap 1 1 agi_language en agi_callerid 0707100100 agi_context default agi extension 123 agi priority 1 Example 4 Example of AGI script An AGI script is composed of a list of variables and values The first variable and value defines which script should be run when the dialplan call this AGI script The rest define what environment should be used when the script is run Therefore you can use a general programming language to do advanced logic manipulate a database or communicate to another server via the Internet Using such scripts is more flexible and powerful than using a static dialplan However using a
35. cellular network The result is that the correct call might not be set up unless there is some means to improve the reliability of signaling the callee s number to the IP PBX Another problem is validating user Currently the IP PBX validates the user based upon the incoming caller ID This feasible because the IP PBX already knows all the numbers of all of the valid users Hence it compares these numbers to the incoming caller ID If the incoming caller ID matches one of these numbers then this user has the right to continue to place a call However some mobile operators do not transfer the caller ID to the called destination or there is an extra charge for this service An example of this problem is Companhia de Telecomunica es de Macau S A R L CTM 29 which is a mobile operator in Macau If the IP PBX does not get a caller ID from the caller then this user will be treat as an invalid user and the call through attempt will be rejected An additional problem occurs because the caller ID of the caller can be faked thus there is a risk that the IP PBX can be fooled into making calls for an illegitimate caller The traditional solution to the problem of not having the caller ID or a faked caller ID 1s to force the caller to enter their account number and a password to place the call through call While this approach is widely used in practice it requires a lot of extra work by the caller This leads to an alternative way of verifying the legitimac
36. eeds to create an account in the DoD server The DoD server will automatically change the appropriate settings of Asterisk 41 Chapter 5 Evaluation This new DoD solution will be released as a commercial product to replace the existing DoD solution The functionality and performance of the system are very important for users In order to quantify the system s performance and to identify weak points that can be improved in the future we tested several aspects of this version of our solution The following sections will explain in detail some of the tests that were conducted 5 1 Test Environment The test environment includes the test machines their operating systems the software network condition and hardware SIP phones Specifically the test environment consisted of e Test machines E Desktop computer CPU Inter Celeron 2 53GHz Memory 1GB Hard Driver 400GB SATA Operating System Fedora 9 Kernel Linux 2 6 27 E Desktop computer CPU Inter Pentium 2 8GHz Memory 1GB Hard Driver 80GB SATA Operating System Windows XP Professional Service Pack 3 Network configuration One of the test machines which was the DoD server This computer has a public IP address Another test machine is behind a NAT on a private IP network The router of this second machine is connected to the same Internet Service Provider ISP as used by the first test machine See figure 24 Software E Web Container and Application Server JBoss Applicati
37. ers but the response time of each page is almost three seconds so the users perception will be that the system 1s a bit sluggish although it is still normal However the user must wait for a long time which is not be accepted For example after the test has been running for roughly 35s thus all 200 users are in the system the response time to get the user s address book page is on average 10 44s this will be perceived as very bad performance Hence the system cannot successfully support 200 simultaneous users with the utilization model that we have used for this testing These test results suggest that the system could support roughly 150 160 simultaneous users with less than 5 second 45 response times Unfortunately this means that we will only be making 75 utilization of our SIP trunk Hence the server s performance needs to be scaled up so that we could server all 200 users while providing less than 5 second response times a mmm Be ow E amp E QA HS A A Measurement Trends standardized values a x x ri a FA 3 mmn Lal bo Po to mo ir Er cn cn Ga Ga J mo Oo no oo Elapsed scenario time ss address book T gt address manager T call history T 2 login_T 2 logout_T gt settings T 4 statistics T Color Measurement Ma inirnuirn Average Maximum Median Std Deviation DY address book T 0 493 5 388 10 444 5 496 2 29 address manager T 0 264 4 630 5 94 5 123 1 964 D
38. es an outgoing call to the callee s number After the callee answers this call then the IP PBX bridges these two calls together When a user places a call back call a HTTP or HTTPS request which includes the caller s number the callee s number the username and a password is sent to the DoD server The DoD server creates and sends a call request to Asterisk After Asterisk receives the request sent from the DoD server Asterisk places two calls one to the caller and another one to the callee After these two calls are both answered Asterisk bridges these two calls together If the caller or the callee is not answered after 20 seconds the call will be terminated As mentioned earlier using a database is an important part in Asterisk In the IP PBX databases have been used in several places Currently three databases are used 1 CDR By default Asterisk generates CDR records as comma separated text files in the var log asterisk cdr csv directory The file master csv contains all the records Detail records can be partially configured on a per channel basis and some of the data for IAX and SIP can be determined on the user level The Zap configuration for CDR records is determined by the configuration of the ZAP channel in zaptel conf 30 In order to make it easy for other applications we save all CDR information in a MySQL database The following information has been stored in the database start calling date duratio
39. es the type of user There are three types peers A SIP entity to which Asterisk sends calls a SIP provider for example users ASIP entity which places calls through Asterisk A phone which can only place calls 19 friends Botha peer and a user hosts Defines the location of this SIP entity This could be an IP address domain name or dynamic If dynamic 1s specified then the location of the host will be specified when the host registers with the SIP registrar secrets The password of the SIP entity to be used for authentication t You can find this file in the directory etc asterisk Dialplan The dialplan is truly the heart of any Asterisk system as it defines how Asterisk handles inbound and outbound calls 20 The configuration of the Dialplan is specified in the file extensions conf There are several important concepts used in this configuration file two of the most important are Contexts and Extensions E Extensions In a traditional telephony system the extension represents a specific telephone line However in Asterisk an extension could be more general An extension is a series of steps in a context If a call matches a certain extension then the call is sequentially processed by the logic defined for this extension Each extension is composed of 3 parts name priority and action The name represents the extension name or number The priority indicates the order of execution with the highest priority operatio
40. esult the callee will always see only a number that can be used by the callee to call user A via the IP PBX This makes the number or SIP URI used to call the callee invisible to the callee all through LJ Mobile 0707100100 B s mobile PBX number 0757008811 CID 0757008811 Local extension 8811 Figure 6 Showing the user s IP PBX assigned caller ID when using call through When user A places a call through call to another user of the same DoD system in this case user A simply called user B s local extension then user B s device should display the caller ID as user A s local extension If user B has enabled their followme feature then the followme phone should display the caller ID as user A s IP PBX number See figure 7 22 all o e Mobile 0707100100 OCs exe aston PBX number 0757008811 p CID 8811 Local extension 8811 a Followme number CID 0757008811 Figure 7 Showing different caller IDs when using call through between two users connected to the same DoD system 3 2 2 Caller ID when using Call back When user A places a call back call to someone s mobile number then user A will first receive a call with the caller ID of user A s IP PBX number because user A initiated this call After user A answers this call then the callee will receive a call with the same caller ID See figure 8 A bac k LKJ Mobile 0707100100 Someone s mobile PBX number 0757008
41. f context 2 2 1 2 Database configuration In Asterisk a database is used in two ways The first is for storing Asterisk settings This is called the Asterisk Realtime Architecture ARA The second use of a database is for storing call detail records ARA stores the Asterisk configurations While the configuration could be stored in files by using ARA you can store the configuration in one of several different kinds of databases There are some advantages of using the Asterisk Realtime Architecture rather than configuration files The biggest advantage is that you can design a distributed Asterisk system The configuration of Asterisk could be stored at another host then a number of Asterisk servers can use this shared configuration to their users information Note that by using a database you also have the traditional database features of roll back to restore the configuration to an earlier configuration and consistency so that the configuration even in the distributed case can remain consistent Additionally applications can manipulate the configuration for example providing a GUI for showing and modifying the set of configurations The Asterisk Realtime Architecture could be Static or Dynamic Static ARA If you use a static ARA to store configurations you have to reload the configurations after you change the configuration Normally a static ARA is used to store configurations which are rarely changed e Dynamic ARA A
42. fficult However if your mobile device has a VoIP user agent and you have suitable connectivity then you can easily use VoIP from your mobile device To provide a flexible solution to VoIP everywhere even to devices that do not or can not have a VoIP user agent Opticall AB has designed Dial over Data DoD solution By using this solution you can place a VoIP call from your mobile device or even fixed phone without requiring that the device that you use have a VoIP user agent This solution also provides a central Internet Protocol Private Branch eXchange IP PBX which can connect call to and from to Session Initiation Protocol SIP phones Both individuals and companies can use this solution for call cost savings Max Weltz created the existing DoD solution in an earlier thesis project This thesis 1 provides a good description of the existing DoD solution As a result of continued testing and user feedback Opticall AB has realized that their DoD solution needs to be improved in several area This thesis project first identified some of the places where improvement was needed explains why these improvements are necessary and finally designs implements and evaluates these changes to confirm that they are improvements An important result of this thesis project was a clear demonstration of improvements in configuration of the solution better presentation of call data records correct presentation of caller ID and the ability to u
43. icense GPL as a free software license and a proprietary software license to permit licensees to distribute proprietary unpublished system components 15 2 2 1 How to configure an Asterisk server 2 2 1 1 Configuration files The configuration of Asterisk is based on configuration files There are many configuration files which are responsible for different settings The two main aspects of configuration are 1 to configure any phones that are to be used with this IP PBX and 2 to define a dial plan this defines the numbering scheme to be used within this IP PBX Configuring SIP phones One of the main reasons for using Asterisk 1s that Asterisk supports the SIP protocol and can utilize SIP phones as IP PBX extensions The configuration of these phones is specified in the file sip conf In this file you can configure both SIP phones and SIP peers Example I shows how to configure a simple SIP phone sip phone 1 type friends host dynamic secret PASSWORD Example 1 Example of a sip conf file There are many fields that can be specified when setting up a SIP phone or SIP peer You do not need to set every field you only need to set fields which differ from the default setting The first line in above example specifies a name for this SIP phone this is simply an identifier but this ID is very useful for an administrator to refer to this SIP phone The following fields are very important for a SIP phone types Defin
44. ified development of distributed transactional secure and portable applications based on Java technology 25 11 2 3 1 1 EJB components The EJB components come in 3 types Session bean Message driven bean and Entities Each type of EJB component is responsible for a specific task These tasks may communicate with each other Session bean Session beans are responsible for business and application logic throughout the whole application They are invoked by EJB clients such as a web browser or normal GUI based application As the name suggests a session bean exists in a session This session could be between the EJB server and client After the session the session bean will be destroyed Normally session beans are invoked by EJB clients to execute some business logic or to invoke other EJB entities to store persistent state There are 2 types of session beans one is stateful and another 1s stateless A stateful session bean can remember state for specific a user until the user completes all of his or her actions In contrast the stateless session bean cannot maintain state So the stateless session beans are normally responsible for independent actions Message driven bean Message driven beans can also invoke business methods but message driven beans are not invoked by an EJB client instead they are invoked by messages Thus when a messaging server receives a signal message a message driven bean can be invoked Entities E
45. ing a incoming call Ideally if the caller is also another user of this same DoD system then the caller ID of the incoming call as displayed at the local extension should be the caller s local extension number This would have the advantage that all calls between users of the same DoD would all appear on local extensions as calls both to and from local extensions Additionally it would mean that a caller could always dial the IP PBX number of the callee but this would be handled as a call between local extensions when both parties were using their local extensions The advantage of 25 this approach is that the caller does not have to know or even think about if the callee is at their local extension or not they simply always dial the same number for this party Note that the reverse is not true 1 e the caller cannot always call the callee at their local extension unless the caller is using the Symbian client or their SIP UA 1 e this would not work with a mobile or fixed phone as the switched attached to the access network would not be able to interpret these local extension numbers since they do not exist in the actual local access network However in some mobile and fixed phone networks there is a facility to define short numbers if there were a short number defined for each of the relevant local extension numbers then a similar behavior could be provided but this would not work with all mobile and fixed phone networks
46. ing f r att minska samtalskostnader Max Weltz vidareutvecklade den befintliga DoD l sning 1 ett tidigare examensarbete Denna avhandling 1 ger en god beskrivning av den befintliga DoD l sning Som ett resultat av fortsatt testning samt synpunkter fr n anv ndarna har OptiCall AB insett att deras DoD l sning m ste f rb ttras p flera omr den Detta examensarbete har 1 f rsta hand identifierat n gra omr den d r f rb ttringar beh vdes f rklarat varf r dessa f rb ttringar r n dv ndiga och slutligen utvecklat och utv rderat dessa f r ndringar Ett viktigt resultat av detta examensarbete visades av en tydlig demonstration av f rb ttrad utformning av l sningen Gr nssnittet fick bla en b ttre presentation av samtalshistorik mer korrekt nummerpresentation Dessa f rb ttringar b r g ra denna l sning mer attraktivt f r de personer som skall anv nda och underh lla l sningen Table of Contents Chanter INTE OCIUCU OT saesapensssserbotaost ssnisins nkedesnn nee NN sins rensa et N reses 1 or BC KE ROU sree ris sedan b n or rd sansad sr oe der cate tesa orovsucaneemntagasseanereatnceccnenpodeessecoonanncee 3 DM SIP OC OC cesses sete cane po stin sov E ET etikens ns ins Nesse 3 2 lds VV Piet Ol CANN IO tite dinseetace A A E E E T 3 e SIE COMMOOINC US e news naos amined nacer tara anoeat eran aaa seenneereesonenet 3 PED SIP UNC SS Ae CS E E E E ates tosa E E EE rr eda ison ideation Canetonnaecnses 4 2 1 4 How
47. initiate a session There are four kinds of severs used with SIP HB Location server Used by a Redirect server or a Proxy server to obtain information about a called party s possible location The location server is used by the Registrar server to store location information Note that the location server is not a SIP server since it does not utilize the SIP protocol Mm Proxy server A Proxy server is an intermediary that acts as both a server and a client for the purpose of making requests on behalf of other clients Requests are serviced internally or transferred to other servers A proxy interprets and if necessary rewrites a SIP request message before forwarding it H Redirect server A redirect server that accepts a SIP request maps the destination address into zero or more new addresses then returns these addresses to the client Unlike a Proxy server it cannot accept calls but can generate SIP responses that instruct the User Agent Client UAC to contact another SIP entity Mm Registrar server A Registrar server accepts REGISTER requests A registrar 1s typically colocated with a Proxy or Redirect server and may offer location services The registrar saves information about where a party can be found 18 Each user agent that will register with this server must have some type of trust relationship with this registrar in order to be able to register deregister and to protect the communications between the two 2 1 3 SIP
48. ith the IP PBX acting as their SIP proxy Users will be charged they place an outgoing call via the IP PBX Users can 20 modify their SIP account by using web GUIs at their DoD server The modified SIP account is connected to user account in the DoD solution because we save the username in the field userfield in each CDR In addition users can place a call to someone s SIP account via the DoD server using call through or call back For instance when using call through user A wants to call the SIP phone of user B from their mobile phone Instead of entering the number of B user A enters the SIP account of user B see figure 4 step 2 Then the IP PBX will place a call to the SIP phone of user B A similar approach can be used when using call back User A sends a HTTP request including the SIP account of user B see figure 5 step 1 then the IP PBX will place a call back call between user A and the SIP phone of user B Currently there is a limitation when using call through to call someone s SIP account from a cellular or fixed phone when the user wants to enter B s SIP account after establishing a call to the IP PBX as the user can only enter digits thus the SIP account you want to call must consist only of digits 3 1 3 2 Incoming calls The users in DoD solution not only place calls but also can receive calls Each user will be assigned a number which can be reached from the PSTN and PLMNs If anyone wants to call user
49. l of this thesis project was to improve the server side of the prior DoD solution The complete DoD solution provides two main methods to place calls via mobile devices Based upon earlier user tests there were some problems with the existing DoD server Additionally we found some other aspects of the existing solution that could be improved In order to provide a more stable and user friendly solution for users we made some changes and improvements in both the IP PBX and DoD server In particular we improved the dialplan By using ARA the dialplan and SIP settings are saved in a database This makes it possible for applications to control and manage Asterisk Additionally changes in settings of Asterisk can be made without reloading Another improvement was optimizing the caller ID by using a local channel and the command NoCDR this enables Asterisk to deliver the correct caller ID to different kinds of devices at the same time At the same time our new configuration of Asterisk can filter call records keeping only the relevant CDR information in the database We have also improved the graphical user interface to show call data records to users and administrators The final major improvement was to rewrite the DoD server using EJB3 making it both easier to write EJBs and reducing the number of lines of code that need to be written and maintained The new improved server is working correctly with the Symbian client this client is the topic
50. laying the HTTP or HTTPS requests sent from each virtual user and checking the responses from the web server There are four steps to testing using LoadRunner I Record the actions of a representative user into a script 2 Replay and verify the script to make sure it can be run correctly 3 Create a scenario with a specified number of users defining the iteration time schedule and so on for this scenario 4 Running this scenario as a test We assume that 25 users of all the users are always calling via the DoD solution Because the SIP trunk that we are using can support 50 simultaneous calls this means that the largest number of users that our tests environment can handle is 200 users 5 2 3 Test results The first part of this test is to make sure the DoD server is stable while supporting a certain number of users in this test 200 users So we conducted a test that simulates 200 users From the user s point of view the system s performance is how long the user needs to wait for each request before getting a response A typical rule of thumb is that if the response time for pages is within 2 44 second then the user perception is that the system is perfect between 2 and 5 seconds is normal and having to wait more than 5 seconds is bad In the following plots Vusers is the number of virtual users Figure 25 shows the number of users in the system as a function of time As can be seen the number of users increases by 10 users pe
51. lloWorldBean remote Example 7 Example of using JNDI EJB private HelloWorldBeanLocal helloWorldBean Example 8 Example of using DI 2 3 2 JBoss JBoss Application Server JBoss is an open source implementation of the Java EE suite of services It is comprised of a set of bundles for enterprise customers who are looking for preconfigured profiles of JBoss Enterprise Middleware components that have been tested and certified together to provide an integrated experience 28 The JBoss runs on a Java virtual machine JVM On top of JBoss are your enterprise applications as shown in Figure 3 15 Users Your Application JSP Servlet EJB Jboss Java EE platform JVM Figure 3 Web application structure in JBoss 2 3 2 1 Authentication of JBoss JBoss provides a feature for authenticating and authorizing users The feature is called a security domain A security domain defines different roles and different resources Each user has one or multiple roles Each resource can be accessed by one or multiple roles We define this security domain in the file login config xml In this file you specify how to authenticate users You can authenticate users based upon a user name and password stored in a file using a relational database and hashed password or even using a LDAP server In a security domain if you use a relational database then y
52. lly a very long timeout can be used for users manually entering this information via the keypad of their phone Alternatively a new JavaME client could be written that directly generated the combinations of the two tones needed for the DTMF dialing thus allowing rapid signaling of the call through number for both clients 6 2 3 Other ways to place call back calls SMS and Voice Currently we send an HTTP request to the DoD server to tell it the caller and callee s numbers and to provide a username and password However in some countries it is expensive to send information via a mobile phone s Internet connection Fortunately there are several 51 alternative for placing call back calls Two of the most obvious are SMS and voice These two alternatives are described below 6 2 3 1 SMS call back Another means of providing the IP PBX with the caller and callee s number password and username would be via a SMS The IP PBX or DoD server could receive these SMSs messages Asterisk can connect to a GSM gateway or a mobile phone or bank of mobile phones could be used to receive SMSs Similarly the SMSs could be received by the DoD server There are many SMS providers that provide a service for receiving SMSs via a web interface As a result the DoD server could check for incoming SMSs by sending a request to such a provider or the provider can send a HTTP POST request including the content of the SMS to us By using these t
53. m appserver index html Last access on 2009 08 20 45 IBM WebSphere www http www 01 ibm com software websphere Last access on 2009 08 20 46 Apache Struts www http struts apache org Last access on 2009 08 02 47 JavaServer Faces Technology www http java sun com javaee javaserverfaces Last access on 2009 08 02 56 TRITA ICT EX 2009 104 www kth se
54. messages In order to explain clearly how the SIP protocol works we use a simple example to explain how it works This example illustrates the traffic exchanged to set up and tear down a SIP session between SIP user agents A and B See figure 1 and table 1 Figure 1 Simple message flow in setting up and tearing down a SIP session between A and B 1 INVITE A sends an INVITE message to invite B to join a session 2 180 Ringing After receiving the INVITE message B starts to ring indicating to the user that there is an incoming call and sends back a 180 Ringing message 3 200 OK After B answers the phone B sends back a 200 OK message to tell A that it is ready to communicate 4 ACK After A receives the 200 OK message A sends back a ACK message and starts to transfer voice content via RTP 6 BYE If B wants to hang up the phone then B sends a BYE message After this B stops sending RTP packets 7 200 OK After A receives the BYE packet A sends back a 200 OK message i then A stops sending RTP packets Table 1 Detail explanations of SIP messages 2 1 4 How SIP works in the DoD solution The DoD solution is designed to provide cheaper calls so SIP plays a very important role in this solution By using SIP the cost of both placing and answering calls can be low cost If a user calls another user and both users are using SIP user agents then these two users do not need to pay for a circuit switched call If a user place
55. n to be performed first The action represents the action to be taken for a call of this type for this extension Example 2 shows an example of an extension In this example the extension has the number 8801 and the highest priority action is to dial while the second highest priority is to hangup Thus after the call is completed successfully or unsuccessfuly the extension will be hungup exten gt 8801 1 dial SIP 8801 exten gt 8801 2 Hangup Example 2 Example of an extension Contexts Dialplans are broken into sections called contexts Contexts are named groups of extensions which serve several purposes 20 These contexts can be used to split the logics for handling calls For example one context might only be for outgoing calls while another context is only for incoming calls Example 3 shows an example of two contexts The first of these contexts is used for calls to the number 8801 while the second is for calls to the number 8802 Note that the names of the context are purely for human consumption the Asterisk system has no understanding of these strings 1 e from sip 1s a name of a context used for the convenience of the administrator who defines the dialplan and does not actually mean anything to Asterisk from sip exten gt 8801 1 dial SIP 8801 exten gt 8801 2 Hangup incoming calls exten gt 8802 1 dial SIP 8802 exten gt 8802 2 Hangup Example 3 Example o
56. nality After the first script has successfully be run then we will use a testing tool called LoadRunner 38 to simulate several users in order to test the overall load that the DoD server can support Following this testing we analyze the results As mentioned in section 5 1 the test machine where we put the DoD server has a public IP address We installed LoadRunner on another physical machine There is an Internet connection between these two machines The LoadRunner emulates multiple virtual users Each of these virtual users will execute a script we previously recorded As a result of emulating multiple users many requests will be sent to the DoD server and the LoadRunner will record the results for our later analysis 5 2 2 Test tool HP s LoadRunner is set of applications that provide businesses a way to reliably simulate the load of multiple users against their given application LoadRunner emulates hundreds of thousands of concurrent users to apply production workloads to almost any client platform or environment LoadRunner stresses an application from end to end applying consistent measurable and repeatable loads It then uses the data to identify scalability issues that can affect real users in production 39 The LoadRunner can be used for testing different kinds of applications on different platforms In our case we use 1s to test a web application The LoadRunner tests the web application by recording and rep
57. nd PLMN General users can call other SIP users for free or can call users attached to the PSTN or PLMN at a low cost or even free However these VoIP user agents can only be used on computers or a few smart mobile phones Only Nokia has its own SIP user agent inside their Symbian platform There are some free SIP user agents running on mobile device such as Fring 13 However you cannot place a call from a mobile phone or fixed phone without a SIP user agent Opticall AB sells a solution called Dial over Data DoD which can solve this problem The DoD solution provides a low cost telephony service for mobile phone users fixed phone users and users using a SIP user agent By using the DoD solution a user can place a VoIP call in several ways even placing a VoIP call from mobile phone or fixed phone without a SIP user agent by splicing calls together The complete solution consists of an IP PBX Asterisk 14 and mobile phones with an optional Symbian client The IP PBX is connected to an upstream VoIP provider who is connected to one or more PLMNs so the IP PBX can allow the user to place a call to or receive a call from these PLMNs of course the VoIP providers are also connected to one or more PSTN operators so calls can also be delivered to or received from fixed phones The users can use a SIP user agent a mobile phone or even a fixed phone to place low cost calls In this solution as it does not need to use a SIP user agent at the user
58. ng The existing DoD solution is controlled by two different web applications One is used for controlling the DoD server and the other is FreePBX 36 for configuring the IP PBX Asterisk This is inconvenient for the administrator For example if the administrator wants to add a user to the DoD sever then the administrator has to create an account in the DoD server for receiving requests from this user and create a SIP account in the IP PBX by using FreePBX then the administrator needs to specify the inbound route for this user Even worse if the administrator needs to enable or disable the followme feature for this user and they could not manually change this configuration using FreePBX However the user might want their followme setting changed frequently Another problem with using separate web applications to control the DoD solution and the IP PBX is that FreePBX has its own format for configuration files If we want some feature but FreePBX has not implemented this feature we cannot manually configure those configuration files If we change any settings of Asterisk by using FreePBX then we have to reload the whole Asterisk configuration to make these settings take effect We can save all the settings in database but FreePBX does not support saving settings in a database For these reasons we created a single GUI for both placing calls and configuring Asterisk For example if an administrator wants to add a new user he or she only n
59. ng and terminating two party unicast or multiparty multicast sessions consisting of one or more media streams 6 The details of SIP will be explained in section 2 1 Since the encoded voice stream is transferred through an IP network individual calls are very inexpensive sometimes even free because Internet service is generally a flat rate service and the number of bits or packets per second required to carry voice is quite limited in comparison to other types of traffic particularly large files images and video VoIP is increasingly popular because the cost of a VoIP call between two points is lower than placing circuit switched calls via the Public Land Mobile Network PLMN or Public Switched Telephone Network PSTN Additionally because many VoIP providers provide connectivity between their VoIP network and the PSTN and PLMN a VoIP user is able to place calls to both fixed and mobile phones Depending upon the location of the voice gateway and the agreement of this VoIP provider with these other networks this cost can vary The easiest way to place a VoIP call is using a computer This is because that there are many VoIP program such as a SIP user agent or other VoIP user agent available to use Example of such application are MiniSIP 7 x lite 8 and SJPhone 9 There are also a number of SIP service providers such as Nonoh 10 VBUZZER 11 and VOIP STUNT 12 These service providers are interconnected with both the PSTN a
60. ng local channels to create multiple contexts when there is more than one destination for a call To implement this we simply need to insert the correct information into the dialplan database table The procedure for adding or modifying an entry into the dialplan table can automatically update the correct rows to realize our rules 4 3 Improving CDR records In the existing DoD server there were several web GUIs for statistical information for example showing only the duration of last call the duration of today s calls the duration of this week s calls and the duration of this mouth s calls This information was not useful to the user Users may want to know their call history in detail including call date destination number caller number and so on In addition an administrator might want to check the call history for one or more users In order to provide more information to both users and administrators we create several new web GUIs to show detail call history User can get the following information from this GUI see figure 20 caller s number callee s number call time duration Since the administrators may need to see all users call history We created another GUI for the administrator see figure 21 Using this GUI administrators have an overview of all users If the administrator wants more detail information about a specific user they can selected this user and see all of this user s call history 36 C
61. ns source number destination number and so on We also save a username for each record so we can identify who has placed each call 2 Dialplan and ARA The configuration file extensions conf contains the dial plan of Asterisk the master plan for control or execution flow for all of Asterisk s operations In particular the configuration controls how incoming and outgoing calls are handled and routed This is where you configure the behavior of all connections through your PBX 31 In the dialplan the logic specifies how to route outgoing and incoming calls One important aspect in the dialplan is how to display caller ID to different devices as some calls will lead two different kinds of devices ringing at the same time The need to wait could be eliminated if a distinguished toke was used to terminate the dialed number for instance the symbol IAX is the Asterisk exchange protocol that can be used for VoIP service within and between Asterisk exchanges T Zaptel interface cards are used to interface to traditional circuit switched telephone equipment both telephones and exchanges 27 However Asterisk only supports one caller ID for each call After much effort we found out it is possible use a LOCAL channel to solve this problem LOCAL is an Asterisk pseudo channel Use of this channel simply loops the calls back into the dialplan in a different context Looping the call back is useful for recursive routing as it is
62. ntry in the database Additionally Asterisk validates the users by checking the database Therefore out next test is designed to test the performance of database 5 3 1 Test plan Because LoadRunner cannot test MySQL we use another test tool Apache s JMeter details of this tool are given in the next subsection As mentioned earlier this DoD server was expected to handle roughly 200 users In the previous test we measured the performance when accessing each of the 7 different pages that a user would normally use We assume that each page requires 2 database queries and each page will be required twice Because that in almost all pages the information showed to users may be not in the same table of database thus we normally need to query database to get all data we need Therefore in this test we measure the effects when 200 users each have to query the database 4 times 5 3 2 Test tool Apache s JMeter is open source software It 1s a 100 pure Java desktop application designed to load test functional behavior and measure performance It was originally designed for testing web applications but has since expanded to other test functions 40 Just like HP s LoadRunner we can define users which are called a thread group in JMeter Next we specify one or more JDBC requests to perform for each of this thread groups The final element of the test plan is a Listener The listener is responsible for these all of the results of these JD
63. oOp Find a followme number for local extension S EXTEN from user SiC DRfaccountca 32 followme as 2 Set CALLERIDinum S ODBC_FIND_INCOMING_NUMBER_BY_USERNAME SICDR 55 followme x 3 Set CDRiuserfield bill 39 followme x 4 dial SIP sting S ODBC_FIND_FOLLOWME_NUMBER_BY_EXTENSION S EXTEN 40 followme x 5 hangup 75 from sip 1000 1 dial SIP sting 0704648198 77 from sip 1000 2 noop HANGUPCAUSE s HANGUPCAUSE DIALSTATUS 8 DIALSTATUS 76 from sip 1000 3 hangup 66 from sip x 1 NoCDR 5 from sip x 2 Answer 43 from sip x 3 Set EXTEN_LENGTH S i0DBC_FIND_EXTENSION_LENGTH 44 from sip K 4 Set CALLING_NUMBER_LENGTH S LEN Si EXTEN 45 from sip K 5 Gotolf S S CALLING_NUMBER_LENGTH SiEXTEN_LENGTH 78 6 46 from sip X 6 dial LOCAL HEXTEN HEoutgoing calls n 4 from sip hangup 23 from sip x 8 Gotolt S S ODBC_FIND_FOLLOWME_STATUS_BY_EXTENSION S EXTEN O 79 11 4 from sip x 9 dial LOCAL HEXTEN h local extension n 25 from sip x 10 hangup 42 from sip x 11 dial LOCAL S EXTEN H amp local extension n amp LOCAL S EXTEN H follawme n 6 from sip x 12 hangup 42 incoming calls 075700838193 1 System root voicecallback voicecall sh CALLERID num 21 incoming calls 0757008819 hangup 26 incoming calls _xX 1 NoCDR 7 incoming calls _X 2 Noop check whether the number EXTEN belongs this PBX digit timeout is TIMEOL 82 incoming calls X 3 Gotot S SODBC CHECK INCOMING NUMBER BELONGS PBEXK HEXTENDI 0 74 5 Figure 19 Dialplan stored as a table in a database
64. of another master s thesis However there are some aspects of the system that could be improved in the future These changes are described in the following section 6 2 Future work Today the new DoD server is up and running but it is not prefect and still has some problems For example a caller can cheat the IP PBX by faking its caller ID the caller may have to wait a long time to place a call and so on Although there are many areas for future work to improve this new DoD solution in the following subsections we will describe some of the most urgent areas for improvement 50 6 2 1 Using a framework in the presentation layer The presentation layer of the DoD server is composed of many JSP pages However maintaining a web service is only feasible when the web application does not contain many pages However in the future the DoD server may include even more logic and pages thus it would be better to use a framework to manage and control the complete presentation layer Presentation frameworks such as Struts 46 and JSF 47 are well designed for implementing the presentation layer These are already useing a model view controller model 2 architecture and these frameworks can help the developer reduce the amount of logic needed in the presentation layer and provide internationalization support This internationalization support is particularly important for the product as one of the major uses of the DoD solution is with DoD serve
65. on Server 5 0 1 GA E IP PBX Asterisk 1 4 24 1 E Java Runtime Environment 1 6 0 14 E Testing tools LoadRunner 8 1 Feature Patch 4 Jakarta JMeter 2 3 4 Hardware SIP phones HB Hitachi Cable Wireless IPC SOOOAE WiFi SIP Phone 42 Gs Router with NAT Test machine DoD server Figure 24 Network topology of test environment 5 2 Tests for JBoss Application Server DoD server The DoD server is a web application running in a JBoss application server This DoD server is used for processing request from users showing GUIs and responding to user interface commands and sending requests to Asterisk to place calls A key performance metric for the DoD server such is how many users it can handle simultaneously or how many requests can process per second This performance metric is very important for both users and developers Additionally understanding the performance of this new implementation will help us to analyze where we could improve the system in the future 5 2 1 Test plan In order to make this test as realistic as possible we record the actions of a user in a script This user script will do normal actions as a form of smoke testing Smoke testing is a shallow and wide approach to test an application The test script touches all areas of the application 43 without getting too deep 37 Because this smoke testing tests all the main functions of the system we get only a general impression of the system s functio
66. opology of test environment sssssssssssssesserrrrrrrrrerrrrrrrrrrrrrrrrrrrr rr rr rr rr rr nr rn nn rann 43 Feee RU SCT SCA Csssmmbisstemstetsisneietonsnmnonsdisninonstsnnn nen ut ene NS 45 Figure 26 Average response time for each page in SeCONC sssressersrrsrsssrrrrrrrrrrnrrrsesssrrrrnrra 46 Figure 27 Response time to query the database cccsssssccccccceeeeeeeeesseeeseesennneeeeeeeeeeeeeeeseeeeeeeeas 48 VI List of Examples Example 1 Example of a sip conf file ssmsssssssssrrrrsrrrrrererrrrrrrrrrrrrrrrrrrrrrrrrrrrr rr rr rr rr rr rr rr rr rr rr rr rr rr nn rn nn nn nn 7 Example 2 Example of an extensi n eraciren ds deme eeii dew e eii e e eia i 8 Example 3 Example Ol come irene O 9 Example 4 Example of AGI script ssnnnsssseeeeeeeerererererererrrrrrrrrrrrrrrrrrrrrrrrr rn rr rr rr rr rr rr rr nr rr rr rr rr rr rr rn rn nn nn nen 10 Example 5 Example of declaration using XML using annotation ssssseeerrerrrrrrrsrsrssssssssnrrrrre 14 Example 6 Example of declaration sssssssssessererrrrrerrrrrrrrrrrrrrrrrrrrrrrrrrrrrrr rr rr rr enn rr rr rr rr rr rr rr rr nr nn nr nn nn 14 Example 7 Example of using INDI issssms sesiss essies s sen ai 15 Example 5 Example OT eho 10 2B pene bi ope ee Po re a a RN 15 Example 9 SIP account setting in ASterisSk ssssssssssssssrsssrrrrrrrerrrrrrrrrrrrrrrrrrrrrrrrrr rr rr rr rr rr rr rr rr rr rr nn nn nen 34 Example 10 Dialplan saving in flat filQ
67. ore Since the DoD server Asterisk and the database are running in the same machine the database queries are not actually send out on the network as Ethernet frames Hence we have not measured the performance that would be experienced if these servers were actually running on different computers Adding this additional delay to each of these queries would increase the delays by one or more milliseconds even with a 100 Mbps ethernet between the machines Comparing this with the 2 2 milliseconds average time per query for 2800 queries in 6 216 seconds we see that an increase of one or two milliseconds for the network processing could double this average time Additionally comparing this average time over the whole period with the average time delay for the actual queries we see that the distribution of query times has both a long tail and is more than 20 times the simple average over the whole period 48 5 4 Evaluation of Asterisk and the SIP trunk Asterisk also queries the database when checking the caller ID or finding out which caller ID should be presented This also contributes to the load on the database The SIP trunk that we have used for testing supports 50 calls at the same time Thus Asterisk could potentially have to check the call IDs of 50 simultaneous calls and it must also retrieve the correct caller IDs from database to present to users for all of these calls Each call will generate 2 database queries As a result there are
68. ou can also define an SQL sentence to be used by JBoss to determine whether the password is correct and what role the user has After defining the security domain the programmer has to define which resources should be accessed by which roles By using a security domain in JBoss you do not need to explicitly write any code for authentication or authorization 16 Chapter 3 DoD solution As mentioned earlier Opticall AB provides a solution called Dial over Data DoD which can provide a low cost and flexible solution for placing a VoIP call Opticall currently use SIP as the protocol to establish VoIP call sessions By using DoD users can place calls from a computer mobile phone or fixed phone while taking advantage of VoIP when possible No matter whether you have a SIP user agent or not you can place low cost calls In practice the typical user has no idea that they are placing a VoIP call they simply want to place a call One of the goals of DoD is that it should allow the user to simply place the call while DoD hides the details of doing this at low cost Unfortunately this was not the case of the existing solution hence the need for at least one of the improvements presented in the next chapter of this thesis However before we present the problem and its solution we need to introduce the current DoD solution in this chapter The DoD solution includes three components optional Symbian clients DoD server and IP PBX
69. program means that the time to process a dialing event is no longer a fixed delay but can be quite variable depending on the complexity and run time behavior of the program that is executed 10 2 3 Java EE Java Platform Enterprise Edition Java EE builds on the solid foundation of Java Platform Standard Edition J2SE and is the industry standard for implementing enterprise class service oriented architecture SOA and next generation web applications 22 The Java EE platform provides a number of services beyond the component hosting of Servlets Java Server Pages JSPs and Enterprise JavaBeans EJBs Fundamental services include support for XML web services transactions and security 23 Since Java EE 1s based on the Java platform you can use any Application programming Interfaces APIs which belong to J2SE The GUI for these applications could be created using Swing or Abstract Window Toolkit AWT or by a HTML based GUI By using Java EE the applications can utilize the underlying platform to provide security database manipulation and transaction control There are additional advantages of using Java EE Your Java EE application can run in any Java EE application server which supports the Java EE standard That means you are not tied to a specific vendor of a Java EE application server According to http java sun com javaee overview compatibility jsp there are 13 Java EE 5 compatible implementations from a total of 12 organiza
70. r Similar clients could be implemented for other platforms for example Android iPhone OS Windows Mobile and Linux Additionally users may want to place call back or call through calls without a Symbian client for example from non Symbian mobile phones and fixed phones For call through user must manually dial the access number of IP PBX then dial the callee s number For call back on a phone equipped with a web browser the user can browse to a web page for the DoD server then enter numbers and click submit to place a call back call 3 3 2 IP PBX Asterisk The IP PBX is used to place calls via a SIP trunk process HTTP requests from clients and 26 validate users We assumed that the IP PBX is connected to a SIP trunk which is connected to PSTN and PLMNs so that this IP PBX can place calls to PSTN or PLMNS and also receive calls from PSTN or PLMNs When user places a call through call the IP PBX will first validate the caller ID of caller This requires that all of the mobile number of users have already been saved in the database If the incoming caller ID is the same as one of the numbers associated with a DoD user the IP PBX treats the caller as a valid user After the IP PBX answers this call it provides a new dial tone to user Then user sends DTMF tones using Symbian client or manually to indicate the callee s number After waiting several seconds to make sure the user 1s done inputting a number IP PBX plac
71. r second after an initial start up time roughly 5 seconds Each user s call last for approximately 10 seconds As can be seen in this figure after making a call the user leaves the system After 115 seconds all users have made calls and have left the system Note that the values shown as diamonds in the plot are from the data which was collected every tenth of a second This will also be the same period over which results are averaged in the next subsection 200 180 160 140 120 100 a0 Number of users EO 40 20 0 00 00 00 10 00 20 00 30 00 40 00 50 01 00 01 10 01 20 01 30 01 40 01 50 02 00 Elapsed scenario time mm ss Running users Figure 25 Running users chat 5 2 4 Analysis of the test s result s In this test no errors occurred This means the DoD server was stable while handling 200 users with the 25 active ratio and the average call duration of 10 seconds In figure 26 each curve represented the average response time in seconds for each individual page utilized by a user when using the DoD sever The averages are computed every tenth of a second The minimum value of the average response time of each page is 0 12s this is the response time when the user is logging out The maximum value of the average response time is 5 38s when users request their address book page The average response time for all pages is 2 84s This test indicated that this configuration of the DoD server can serve about the 200 us
72. ransform this call history into some other form or mine this call history for information As an example of the first the user might want to get their call history and automatically generate 52 a list of contacts that they have not called within the last month in order to keep up to date with their customers As an example of the later a user might want to have an application that automatically generates entries in the user s calendar to call those persons who called them when they were unavailable 53 References 1 Max Weltz Dial over Date solution Master thesis Royal Institute of Technology KTH School of Information and Communication Technology Stockholm COS CCS 2008 2 Sweden February 2008 http web it kth se maguire DEGREE PROJECT REPORTS 080221 MaxWeltz ExjobbReport with cover pdf 2 VOIP wikipedia www http en wikipedia org wiki VolP Last access on 2009 5 25 3 Skype www http www skype com Last access on 2009 08 02 4 Philippe Biondi and Fabrice Desclaux Silver Needle in the Skype BlackHat Europe March 2nd and 3rd 2006 http www secdev org conf skype BHEU06 handout pdf 5 RTP wikipedia www http en wikipedia org wiki Real time Transport Protocol Last access on 2009 5 25 6 SIP wikipedia www http en wikipedia org wiki Session Initiation Protocol Last access on 2009 5 25 7 MiniSIP www www minisip org Last access on 2009 5 25 8 X lite www http www counterpath net Last
73. re simply annotations 34 So by using annotation you can annotate your code rather than 38 writing XML descriptors in another file see figure 22 package com dod jpa import java io 5erializable L ontect impi ments Serializable private oo QS opm y T mh a hh aa private User username Private static final long s2 riaiVersionvID Figure 22 Annotation for EJB3 4 4 1 3 Using Dependency Injection instead of JNDI look up When using EJB 2 if you want to have an instance of an EJB you have to use JNDI look up to get an instance This is a very repetitive operation for the developer In EJB 3 you can use Dependency Injection to get an instance You simply write an annotation above the class members When the class is initialized the container will automatically get an instance for this class member See figure 23 39 package com dod ejbh private UserManacerBbeanLocak userManagerbeaen Figure 23 Dependency Injection for EJB3 4 4 2 Using model view controller mode 2 The structure of the existing DoD server was roughly two layers The first layer is a presentation layer which is used for showing GUIs the second layer was an EJB layer used for all logic and database manipulations This is basically model view controller model 1 architecture Model 1 is the most common way in the web application development from the emergence of the web Entry point for every request starts from
74. rough method of placing a call is essentially the same as the widely used approach to making low cost calls through a local access number By using call through to place a call the user has to place a call to the IP PBX which will normally be in the same country as caller 1 e this will ideally be a local call Subsequently the IP PBX has to place another call to the callee thus the cost is the sum of the cost of these two calls The sum of these two calls can be lower than the cost of a direct call in several cases For example if the IP PBX is within the local calling area of the user then the call to the IP PBX might be very low cost for example in some calling plans this might only be a fixed opening charge and no per minute charges flat rate or some other charging scheme The call from the IP PBX to the callee can take advantage of the IP PBX s connection to a SIP trunk a multiplexed 18 leased trunk the volume calling prices of the IP PBX etc If a distributed IP PBX is used for example in the case of a multinational firm then the traffic between the IP PBXs can be sent cheaply and the call connected to the callee might be a local call However there are also a few problems about call through The first concerns sending DTMF tones DTMF tones are set over a cellular network as audio tones when received at the IP PBX side the IP PBX must decode these audio tones However the DTMF tones can be lost or distorted by the
75. rs located in multiple countries in order to reduce the cost of roaming 6 2 2 Multiple access numbers for the IP PBX The current IP PBX used in the DoD solution has only a single access number for call through no matter what kind of client is using this number When placing call through calls the client needs to send DTMF tones but the speed of sending DTMF is different in different kinds of clients For example in the Symbian client the speed of sending the necessary information via DTMF can be very fast However when sending DTMF tones using a client written in JavaME the client send a request to OS of the mobile phone then the mobile phone itself sends these DTMF tones for client As a result the client cannot control the speed of sending DTMF tones so different clients take different amount of time to send the required information However when the IP PBX is receiving a call through request the IP PBX uses a timeout to decide whether the client has finished sending the required information If the IP PBX only has a single access number then the timeout must be set to be the time required for the slowest client but a user with a faster client will wait unnecessarily in order to place call through calls One solution is to use multiple access number for the IP PBX For example a very short timeout can be used for the access number used by Symbian clients a long timeout at another access number for Java clients or other clients Fina
76. rt Messaging Service Service Oriented Architecture Transport Layer Security User Agent Client Voice over Internet Protocol Virtual Private Network Chapter 1 Introduction VoIP is a general term for a family of transmission technologies for delivery of voice communications over an Internet Protocol IP networks such as the Internet or other packet switched networks Other terms frequently encountered and often considered synonymous with VoIP are IP telephony Internet telephony voice over broadband broadband telephony and broadband phone 2 Note that there are subtile distinctions between these terms but for the purpose of this thesis we will consistently use the term VoIP In order to transfer voice over an IP network several protocols are widely used SIP H 323 and Skype 3 SIP is the most popular and widely used protocol Both SIP and H 323 signaling are used in combination with the Real time Transport Protocol RTP for transferring the actual media session In contrast Skype is a proprietary protocol of which relatively few details are know most of these by reverse engineering 4 The Real time Transport Protocol RTP defines a standardized packet format for delivering real time audio video and timed text over the Internet 5 SIP is a signaling protocol used for setting up and tearing down multimedia communication sessions such as voice and video calls over the Internet The protocol can be used for creating modifyi
77. s a call to PSTN or PLMNs in order to reduce the cost the DoD solution uses a SIP trunk to place these calls So the cost of placing such calls can also be cheaper than a single circuit switched call If a user answers a call which is redirected by the DoD server then the user will be charged for this call However if this call is also placed using a SIP trunk then the cost of answering this call is also cheaper as the call can be delivered to a local gateway Of course the operator of the user s phone may also charge the user for making or receiving the call It is important to note that the cost of calls using a SIP trunk depends on which SIP provider we are using One unfortunate side effect of this method of reducing the cost of a single call by turning it into two or possibly three separate call legs is that the call detail record now consists of two or three related records rather than a single record see figure 2 This leads to increased call detail record processing this will be addressed in section 4 3 Fol lowme Calls Followme number First call DoD server Second cal Callee Figure 2 Multiple call detail records when placing calls 2 2 Asterisk Asterisk is open source program Asterisk can be used as an IP PBX a media gateway and or a call center Asterisk supports several VoIP protocols including the SIP protocol Asterisk is released under a dual license model using the GNU General Public L
78. s to manage users Administrator Current user optical Role Administrator Users Statistics Settings PEX Number IP address of Asterisk Username Password Password repeat Length of extension DISA timeout Figure 16 The web GUI for administrator to manage the connection between Asterisk and DoD server 30 Current user opticall Role Adminis Users Statistics Settings PBX Number OTSTOOsS 1 1 0757008812 0757008813 OTSTO08814 0757008815 0757008816 OTSTOOss 1 T 0757008818 0757008819 Sided Range Add F Set Access Number Figure 17 The web GUI for administrator to modify the numbers of IP PBX 31 Overview chang 32 seconds 0 seconds 13 minutes 34 seconds 0 seconds lopticall 27 seconds 0 seconds 27 seconds 0 seconds Detail Information Username Call Source Call Destination hangi 0707539452 0704648198 Isuntao 0757008813 8812 Start Time Duration 2009 07 21 10 24 03 0 seconds 2009 07 21 10 28 49 0 seconds lsuntao 0757008813 0703008177 2009 07 21 10 28 49123 seconds 7570 2009 07 21 10 29 39 7 seconds lsuntao 07570 2009 07 21 10 29 38 0 seconds Isuntao 07570088130703008817 2009 07 21 10 38 32 5 seconds Isuntao 0757008813 8812 2009 07 21 10 38 33 0 seconds hangi S812 0703008177 2009 07 21 10 46 05 20 seconds optical 07030081770707539452 2009 07 21 10 47 56 18 seconds Isuntao 8813 fs
79. s1 2 2009 07 21 10 55 5510 seconds l nge opin 0012 sana AT 41 1 2 To a mmama de Figure 18 The web GUI for administrator to show the call history 32 Chapter 4 Improvements The goal of this thesis project was to improve the server side of the DoD solution The existing DoD solution was up and running but there were a number of problems For example the web interface did not show CDR records properly the caller ID was wrong when a call had multiple destinations and multiple inconsistent web interfaces were need for controlling the complete DoD solution The following sections will explain each of these problems and how they were each solved in this thesis project 4 1 Saving Asterisk configurations in database By default Asterisk keeps all configurations as files saving in etc asterisk An administrator can change an Asterisk configuration by modifying these configuration files However the Asterisk system has to be reloaded in order for these changes to take effect For example to add a new SIP account to Asterisk you have to modify sip conf then reload the SIP settings into Asterisk In the DoD solution each user has a SIP account and each user can modify their own SIP account through web GUIs Additionally when adding or deleting users a SIP account will be created modified or deleted As a result the SIP settings will be changed frequently Additionally the dialplan will be changed
80. se a number of different graphical user interfaces with the improve DoD solution These improvements should make this solution more attractive to the persons who have to maintain and operate the solution Sammanfattning Detta examensarbete behandlar f rb ttringar i serversidan av OptiCall ABs Dial over Data DoD l sning som tillhandah lls f r tj nsteleverant rer av VoIP VoIP blir mer och mer popul rt M nniskor anv nder VoIP f r att ringa till sin familj och v nner varje dag Det r billigt s rskilt n r anv ndaren r utomlands eftersom de inte beh ver betala n gon roamingavgift M nga f retag vill ocks att deras anst llda skall anv nda IP telefoni inte bara d rf r att kostnaden f r att ringa oftast r l gre utan f r att bolaget kan ers tta sin traditionella f retagsv xel PBX med en kraftfull dator som har PBX programvara f r att ven ansluta alla anst llda till deras VoIP leverant r VoIP leverant ren kan erbjuda billiga samtal till alla anv ndare som ocks r anslutna via Internet Anv ndarna kan hantera VoIP samtal med en VoIP user agent men hur kan de ringa ett VoIP samtal fr n en mobil enhet eller andra enheter utan VoIP user agent Anv ndare vill kunna ringa billiga VoIP samtal verallt VoIP leverant rer vill erbjuda en flexibel l sning f r att locka och beh lla anv ndare ven VoIP user agent finns utvecklade till m nga enheter som en programvara som k rs p en dator en h
81. the behavior of an EJB can be changed without changing the code You can even use both methods at the same time but XML always has the highest priority Example 5 shows an example of using XML and Example 6 shows an example of using annotation to declare an EJB 13 lt session gt lt ejb name gt HelloWorld lt ejb name gt lt local home gt com opticall ejb interface HelloworldLocalHome lt local home gt lt local gt com opticall ejb interface HelloWorldLocal lt local gt lt ejb class gt com opticall ejb HelloWorld lt ejb class gt lt session type gt Stateless lt session type gt lt session gt Example 5 Example of declaration using XML using annotation Stateless public class HelloWorld implements HelloWorldLocal Example 6 Example of declaration Dependency injection In the previous versions of EJB you had to use Java Naming and Directory Interface JNDI lookup to access an EJB Every time you wanted to use an EJB you needed to write similar code In EJB 3 JNDI lookups have been replaced by simple configuration using metadata based Dependency Injection 27 The difference between these two approaches is shown in the following two examples the first one uses JNDI to access an instance of an EJB 14 another is using Dependency Injection to access an instance of an EJB InitialContext ctx new InitialContext try HelloWorldBeanLocal helloWorld HelloWorldBeanLoca ctx lookup He
82. thers have done of Asterisk performance when connected to a SIP trunk such as 42 and 43 5 5 Conclusions We have tested the DoD server and the database by see that although they are stable enough to support for 200 users the performance of the system is not as good as we might hope However the system is capable of supporting 150 160 users with our scenario parameters Additionally we see that the database performance while sufficient on average is not acceptable for all users Thus we need to examine how this could be improved Further more the performance of the DoD sever is poor as the response time for each page is too long in some cases In the future the performance should be improved by optimizing the code changing the web container such as WebLogic 44 and WebSphere 45 and using a more powerful server The least expensive method of improving the performance of the system to meet the needs of 200 users would be to use a more powerful server as all of the current code could be run Alternatives such as partitioning the services over multiple servers might actually decrease the overall system performance 49 Chapter 6 Conclusions and Future Work The project accomplished the goal for this thesis project but new needs for improvements were found In the first section of this chapter we will summarize our conclusions and in the second section as summary of potential future work is given 6 1 Conclusions The goa
83. tions Another advantage is that you can easily reuse components which are implemented following the Java EE standard Security is a very important aspect of enterprise applications Security for accessing resources is facilitated by using Java EE The security model is based on roles A programmer can define what roles can access what resources This security model is already implemented by the application server You only need to change the configuration files of the application server to define roles and users Java EE also supports HTTPS to secure HTTP traffic You can use a self signed certificate or a certificate signed by a trusted certificate authority As to the security of SIP and RTP the Opticall solution uses Virtual Private Networks VPNs to carry this traffic We will assume that since all of the traffic will be encrypted that we do not have to further consider the security of SIP and RTP Another reason for using VPNs in the Opticall solution is to handle the problems caused when signaling and media having to pass through multiple Network Address Translations NATs Fortunately using VPNs solves both the NAT traversal problem and provides security for the SIP and RTP traffic Details of using VPNs for NAT traversal are described in another thesis 24 2 3 1 EJB Enterprise JavaBeans EJB technology is the server side component architecture for the Java Platform Enterprise Edition Java EE EJB technology enables rapid and simpl
84. ubsection 2 3 1 EJB 1s a type of server side component All the logic and operations on the database are handled by EJBs The existing DoD server used EJB version 2 In 2006 EJB version 3 was released Compared to EJB 2 this version is easier to write and it also offers several improvements Those improvements that are of specific interest for DoD are described in the following paragraphs 4 4 1 1 Simplify writing EJB In EJB verison 2 we need two interfaces and one implement class for an EJB Although some integrated development environments such as JBuilder and Eclipse can generate these interfaces and class automatically it is still a little bit complex for developers when developers need to write many EJBs In version 3 the developer only needs to write one interface and a class This is similar to the usual Java programming thus the developers can write and manipulate these EJBs like normal classes 4 4 1 2 Using annotation instead of XML descriptor In EJB version 2 each EJB needs a behavior definition in a XML descriptor ejx jar xml If you want to change the behavior of an EJB you may have to change both the code and the XML descriptor In version 3 you do not need to have a XML descriptor for each EJB instead you can use annotation to define the behaviors of EJBs Annotations provide information about a program but are not part of the program itself They have no direct effect on the operation of the code as they a
85. unt that you can set each with the value you want to set for this attribute for this account Use of default values simplifies creating an account as you only need to specify non default values When using ARA you still use keys and values to define a SIP account but now the columns are keys and the content of a row in the table contain the associated values When 33 creating the table you can specify default values for some or all columns As a result you only need to specify values for those columns that will have a non default value tammari type friend callerid Tuomas Tammisalo lt 1000 gt username tammari host dynamic secret F regcontext tammari internal regexten 1005 dtmfmode rfc2833 insecure very canreinvite yes nat yes qualify yes context merus sipphone pickupgroup 1 callgroup 1 mailbox 1000 default Example 9 SIP account setting in Asterisk The dialplan could be stored in a flat file etc asterisk extensions conf The dialplan is separated into contexts with each contexts having one or more exstensions with priorities and commands This same information can be stored in a dialplan table in the database with each row specifying an extension Just as with SIP settings by storing the dialplan in a database there is no need to re load the dialplan instead the system will access the relevant parts of this dialplan as it operates When using ARA for dialplan We first create a table for
86. ure 11 Showing caller ID when using SIP user agC0t sssssseseeessrrrrrrerrrrrrreresessrrrrrnrrrrrrsrrrrrrrrn rna 25 Figure 12 Showing caller ID when receiving a incoming Call smssssssssssrrrrsrrrrsrserrrrrrrrrrrnrrrrrrrnnnr ra 25 Figure 13 The web GUL Tor call Dack seriorisnscasstriiesnenia e 29 Figure 14 The web GUI for contact book sssssssssssessessersrrrrsrrrsssssssrsrrrrrrrrrreersrsrrrrrrrrrrrrrrrrr reser rr rn nn na 29 Figure 15 The web GUI for administrators to Manage users ceeeeeeeseeeeesteneeeeeeeeeeeeeeeeeseeeeeees 30 Figure 16 The web GUI for administrator to manage the connection between Asterisk and DoD BS TSI E etna octnn E es aetna ce as aoe ec E E 30 Figure 17 The web GUI for administrator to modify the numbers of IP PBX eee 31 Figure 18 The web GUI for administrator to show the call history sssssssesssrsserersrrsssssssrnnra 32 Figure 19 Dialplan stored as a table in a database uismmosssssssssrrsssssrrrrrrrrrrrrrrrrssrrrrrrrrrrrrrrrrrrrrrrrrrrn nn na 35 Figure 20 Web GUI showing CDR for normal users smssesssssssrrrssssrrrrrrrrrrrrreeessrsrrrrrrrrrrrrrrrrrrrrrrrrnn rna 37 Figure 21 Web GUI showing CDR for administrator ssssssssssssssrrrerrrrrsrrsssssssssrrrrrrrrrrrrrrrrrrrrrrrrrs 37 Figare22 Annotation Tor EIB secetei E 39 Figure 23 Dependency Injection for EJB3 sssssssssssssssrsrsrrrrrrerrrererrrrrrrrrrrrrrnrrrrrr rr rr rr rr rr rr rr rr nr rn nn nn nan 39 Figure 24 Network t
87. using HTTP request Step 2 DoD server causes IP PBX to make two calls to both A and B Step 3b IP PBX alls B and bridges two calls IP PBX Step 3a IP PBX calls A 5 B Figure 5 Call back procedure There are many ways to tell the IP PBX A s number and B s number Using a Web GUI via the DoD web server the user can enter A s number and B s number into a web form and presses submit The web GUI sends a POST request this causes the IP PBX to place the call back calls In addition to the web GUI we could also use Short Messaging Service SMS or DTMF to tell the IP PBX the phone numbers in order to cause the IP PBX to place the call back calls An advantage of using HTTP HTTPS is that 1 the HTTP traffic could use Transport Layer Security TLS thus allowing certificate based authentication of both A and the IP PBX and 2 the IP PBX is calling the caller hence the IP PBX can verify that this is a legitimate caller and that this caller caller A is allowed to call B before placing either call 3 1 3 Other call methods 3 1 3 1 Using SIP user agent Since we use Asterisk as our IP PBX you can connect your SIP user agent to the IP PBX As a result a user can use their SIP user agent to place outgoing calls to PSTN or PLMNs via this IP PBX Of course the user can also use their SIP user agent to place calls to other SIP user agents Each user in the DoD solution will automatically have a SIP account w
88. voice mail exten gt 123 I Answer exten gt 123 n Playback tt weasels exten gt 123 n Voicemail 44 exten gt 123 n Hangup from sip Example 10 Dialplan saving in flat file 4 2 Caller ID The existing DoD server did not show the proper caller ID sometimes especially when a call had multiple destinations For example when user A places a call to a local extension of user B then the local extension number of user A should be displayed at B s local extension However the 35 number assigned by the IP PBX to user A was displayed to B As mentioned in section 3 2 there are a number of different situations that have to be addressed Here we define a number of rules that will be used to decide how to display the caller ID The first rule is to hide the caller s actual number instead we want to only display the number assigned to this caller by the IP PBX The second rule is that the caller ID should be appropriate for the device where it is displayed If the destination is a mobile phone then the caller ID should be the number assigned by IP PBX as this would be the number that the callee could call to reach this caller If the destination is a local extension then the caller ID should be the local extension number as this too is the number that the callee could call using this local extension to reach the caller We can address both of these rules by using the func odbc in our dialplan and usi
89. wo methods the DoD server or IP PBX can get the information necessary to place the call back calls 6 2 3 2 Voice call back There is another way to place a call back call The caller could place a call to the IP PBX the IP PBX would hang up the call Because of the incoming call attempt the IP PBX knows the number of the caller from the incoming caller ID Given this information the IP PBX can place a call back to the caller After the caller answers this call the caller tells the IP PBX the number of the callee by sending DTMF tones or by using voice commands to indicate who is to be called Once the IP PBX knows the necessary information the IP PBX places another call to callee After the callee answers this call the IP PBX bridges these two calls as usual for the call back service 6 2 4 Increasing security for call through Today the IP PBX can be cheated by faking a valid caller ID In order to make call through more secure we could add a password using only digits for each user When the IP PBX answers the call through call the caller could be prompted to enter their password This approach would make call through more security but will increase the delay in initiating a call This will also require a change in the logic for call through for the Symbian and other clients 6 2 5 Providing machine readable call history In many cases it is not a human that will process the call history but rather a program that is going to t
90. y of the caller call back This method will be described in the next section 3 1 2 Call back While call through is often perfect for a caller inside the country where the IP PBX is If a caller is abroad or he she cannot send his her caller ID to the IP PBX then the cost of this call is still high or perhaps the caller cannot place a call through call because of a roaming limit by the mobile operator The second way to place a call in DoD solution 1s called Call back Consider about two persons A and B User A is abroad and wants to call B First user A sends a HTTP request to one of Opticall s DoD servers by using a Symbian client or a web browser see Figure 5 step 1 This HTTP request includes A s number and B s number The DoD server receives this request then causes the IP PBX to place two calls see Figure 5 step 2 The first call is to A After A answers this call then the IP PBX places a call to user B see Figure 5 steps 3a and 3b After A and B both answer their calls then the IP PBX will bridge these two calls The scaling is roughly the same as the call through approach but has the added delay of communicating the HTTP request However the call set up for both calls could be done in parallel while in the call through approach the call setups occur sequentially thus their call set up delays are additive 19 DoD Server Step I A sends A s username and password and B s number to DoD server

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