Asterisk Voice Exchange: An Alternative to Conventional EPBX

Transcription

1 2008 International Conference on Computer and Electrical Engineering Asterisk Voice Exchange: An Alternative to Conventional EPBX Mohammed A Qadeer Department of Computer Engineering, Aligarh Muslim University, Aligarh, India maqadeer@zhcet.ac.in Ale Imran Department of Electronics Engineering, Aligarh Muslim University, Aligarh, India aleimran@zhcet.ac.in Abstract This paper highlights the design and implementation aspects of a VoIP based Asterisk voice exchange, Developing a fully functional voice exchange requires to set up a server based on Asterisk, connecting clients to that server with the help of soft phones and then configuring the softphones with the server. Here in our implementation we have connected the clients to the server with the help of IAX protocols. The first part of the paper contains some introductory concepts about VoIP, followed by Asterisk s internal architecture. In the third part of the paper we discuss about the codecs and protocols used by the packet switching based PBX networks and finally we brush up about the design and implementation aspects. Keywords-VoIP, PBX, IAX, SIP, Asterisk, Trunk. I. Introduction The development of Asterisk based Voice Exchange and its applications needs to take into account the growing complexity with the existing PBX (private branch exchange) networks [1]. A promising solution is the voice exchange based on Asterisk, which works on VoIP [2] (voice over internet protocols) and provides a more promising and flexible solution. VoIP technology originated in about 1995, when hobbyists began to realize the potential of sending voice data packets over the internet rather than communicating through standard telephone systems. The idea is to use the internet as a telephone network with some additional capabilities. Instead of communicating over a circuit switched network, this application allows communication between two parties over the packet switched internet and it was in 1995, when the first internet softphone appeared. Fig1:Diagrammatic View of VoIP This concept allowed PC users to avoid long distance charges, hence great for international calling and a promising solution to the corporate world. In fact one of the main areas of its applications are the callcenters, where the customer is able to communicate with an operator via the internet. Recently there have been proposals in various countries to allow internet telephony, which is indeed a welcome step. This means that people can use VoIP services to connect to the traditional telephone. In this case call travels across the IP network, but terminates at the local phone network, whose owner than charges a fee that is typically the price of a local call [3]. Unrestricted VoIP, which is the basic building block of Asterisk based voice exchange makes it possible for people to use their phones either fixed or mobile to call others in the country and internationally for a fraction of the prices they pay now. all that one needs to make a PC to PC call is a VoIP based software. Besides this, Engg economics also plays an important role in popularizing this new concept. A normal telephonic call requires 64 kbps, whereas the same can be handled an as low as 6-8kbps using codecs such as GSM. Hence the bandwidth requirements will be considerably reduced and one can stuff up, more calls in the same given bandwidth, /08 $ IEEE DOI /ICCEE

2 providing a clear cut advantage. In the future people will stop paying for voice calls just like they don t pay for each they sent or each site they view. Of course a service provider would charge a fee for providing internet services, but everything you did beyond that point wouldn t carry any extra cost.but in a future where everyone will always be connected the convergence several commentators have been talking about will finally become a reality. 2. Asterisk as a Voice exchangee Private Branch Exchange is simply a private telephone network which is used within a organization. PBX systems are large installations able to handle many extensions. PBX systemss are highly programmable and customizable, able to meet the needs and demands of different businesses[1]. Fig 2: A conventional PBX Asterisk is a complete PBX in software written in C programming language and it runs on Linux operating systems. Asterisk does voice over IP in many protocols, and can interoperate with almost all standards-based telephony equipment using relatively inexpensive hardware like for ex PCI cards[4]. Asterisk in fact creates a PBX that rivals the functionalities of traditional telephonee based systems. Asterisk based voice exchange is cost effective, low maintenance and flexible enough to handle all voice and data networking.asterisk does PBX switching, CODEC translation and various other applications like voic , conference bridging, IVR and various others. Fig 3: Overview of Asterisk system 2.1. Asterisk s Architecture Asterisk is carefully designed for maximum flexibility. Specific APIs are defined around a central PBX core system. This advanced core handles the internal interconnection of the PBX, cleanly abstracted from the specific protocols, codecs, and hardware interfaces from the telephony applications. This allows Asterisk to use any suitable hardware and technology available now or in the future to perform its essential functions, connecting hardware and applications. The Asterisk core handles these items internally [5] PBX Switching The essence of Asterisk, of course, is a Private Branch Exchange Switching system, connecting calls together between various users and automated tasks. The Switching Core transparently connects callers arriving on various hardware and software interfaces. Application Launcher launches applications which perform services for uses, such as voic , file playback, and directory listing. Codec Translator uses codecc modules for the encoding and decoding of various audio compression formats used in the telephony industry. A number of codecs are available to suit diversee needs and arrive at the best balance between audio quality and bandwidth usage. Scheduler and I/O Manager handles low-level task scheduling and system management for optimal performance under all load conditions Fig 4: Asterisk s Architecture 2.2. Loadable Module API S Four APIs are defined for loadable modules, facilitating hardware and protocol abstraction. Using this loadable module system, the Asterisk core does not have to worry about details of how a caller is connecting, what codecs are in use etc. Channel API The channel API handles the type of connection a caller is arriving on, be it a VoIP connection, ISDN, PRI, Robbed bit signaling, or some other technology. Dynamic modules are loaded to handle the lower layer details of these connections. Application API The application API allows for various task modules to be run to perform various functions. Conferencing, Paging, Directory Listing. 653

3 Voic , In-line data transmission, and any other task which a PBX system might perform now or in the future are handled by these separate modules. Codec Translator API Loads codec modules to support various audio encoding and decoding formats such as GSM, Mu-Law, A-law, and even MP3. File Format API Handles the reading and writing of various file formats for the storage of data in the file system. Using these APIs Asterisk achieves a complete abstraction between its core functions as a PBX server system and the varied technologies existing (or in development) in the telephony arena. The modular form is what allows Asterisk to seamlessly integrate both currently implemented telephony switching hardware and the growing Packet Voice technologies emerging today[2]. The ability to load codec modules allows Asterisk to support both the extremely compact codecs necessary for Packet Voice over slow connections such as a telephone modem while still providing high audio quality over less constricted connections. The application API provides for flexible use of application modules to perform any function flexibly on demand, and allows for open development of new applications to suit unique needs and situations. In addition, loading all applications as modules allows for a flexible system, allowing the administrator to design the best suited path for callers on the PBX system and modify call paths to suit the changing. 3. Codecs and Protocols Asterisk is quite versatile and has been designed such that it can use a wide range of codecs and protocols. Codecs are the mean with the help of which analog signals can be converted to digital and transmitted via an IP network. Table 1: Various Codecs Used by Asterisk [3] VoIP generally uses two types of protocols 1). Signaling Protocols-for setting up a conversation 2) Media transfer protocols for actual transfer of data, once the connection has been set.[6] To obtain good voice quality across the network we need to minimize the packet losses, jitter, and use echo cancellation techniques.generally used protocols for VoIP based technologies are SIP and H.323[7]. SIP is an application layer protocol, designed to be independent of the underlying transport layers and is used for establishing, managing and terminating a multimedia session.. In fact SIP is a text based protocol just like HTTP protocol and its lightweight approach towards setting up a call is what makes it quite a popular choice for VoIP based networks [2]. Fig 5: SIP call set up IAX is the Inter Asterisk Protocol used by Asterisk and is a alternative to SIP,H.323..Its generally used to enable VoIP connections between servers and between servers and clients.iax is a very robust and fully featured protocol yet its quite simple.. It is agnostic to codecs and number of streams, meaning that it can be used as a transport for virtually any type of data. [5]This capability will be useful as videophones becoming common.iax supports both trunking and multiplexing of the channels on the same link. In fact the basic goals for which IAX was designed were 1) Reduction in Bandwidth used in media transmissions 2) To provide basic support for NAT transparency and 3) To be easy to use behind firewalls. Fig 6: IAX could be easily used beyond firewalls Both SIP and IAX generally show the same efficiency for lower no of simultaneous calls; however as number of calls generally goes on increasing, IAX takes the lead in performance over SIP. 4. Implementing the Asterisk Based Voice Exchange We have installed Fedora 4 Linux to implement Asterisk and begin realizing our voice exchange by compiling the Asterisk and IAX has been used to connect multiple clients with the server. The following commands help us in compiling the Asterisk. cd/root/sam/asterisk make 654

4 make install make samples make progdocs Fig7: Snapshot after the complete installation Once it has been successfully installed we can start Asterisk on the server by running the following commands For setting up a client on IAX client on Asterisk we do the following: ;[imran] ;type=friend ;secret=2222 ;auth=md5 ;host=dynamic ;reinvite=no ;canreinvite=no ;qualify=1000 ;dtmfmode=inband;callerid="imran"<2222> ;disallow=all ;allow=gsm ;context=incoming /root/sam/asterisk-vvvc 4.1. Asterisk s Dial Plan The dial plan is stored in a text file, the configuration file extensions.conf. In this file actions are connected to extensions. Each extension belongs to a context, either the default context or a specific context you create, like incoming sip calls, longdistance outbound PSTN calls, local calls, inter-office calls, etc. Users connecting to asterisk all belong to a specific context (specified in the channel configuration file), which is where asterisk looks for advice on how to handle the calls placed by that user, checking the access rights to expensive lines, with different rule sets for local users and contacts calling from an outside line. /etc/asterisk Contains all of asterisk configuration files and logic information. /usr/lib/asterisk/modules Contains all of asterisk s loadable modules, operating asterisk functionality. Applications, channels and resources are located in this directory. /var/lib/asterisk/sounds Contains all of asterisk s sound files for playback and pre-loaded applications (eg: Voic ). /var/lib/asterisk/agi-bin Contains all of asterisk s AGI scripts and AGI logic Configuring the Files We configure the IAX and the Extensions at the following: IAX: /etc/asterisk/iax.conf Extensions: /etc/asterisk/extensions.conf Fig 8: IAX.conf file Whereas for setting up the Extensions on the Asterisk server we do the following: For editing the Extensions configuration file /etc/asterisk/extensions.conf [inbound-from-iax]; Our context for IAX- clients exten => extension no, priority, application (argl,arg2,...) exten => 1111,1,Dial(IAX2/${EXTEN}) exten => 2222,1,Dial(IAX2/${EXTEN}) exten => 3333,1,Dial(IAX2/${EXTEN}) exten => 4444,1,Dial(IAX2/${EXTEN}) exten => 5555,1,Dial(IAX2/${EXTEN}) exten => 6666,1,Dial(IAX2/${EXTEN}) Fig 9: Extensions.conf file Once we are done with this, we need to concentrate on the installation and registration of the soft phones that we are going to use at the client end. 655

5 ASTERISK PBX Host: Home Analog phones connected to FXS card Phone Line connected to FXO card SIP Home 3001@ hom e 3002@ hom e SIP School 4001@ school 4002@ school ASTERISK PBX H o s t: S c h o o l Analog phones connected to F X S c a rd yet done the connection of our server with the conventional PSTN, though it could be done with the help of PCI cards like for example Digium s TDM400P to validate the connectivity with the existent circuit switched network SERVER A PSTN IN T E R N E T Fig 11: IDEFISK softphones being registered on the server Asterisk PBX currently support 4 types of extension: Zap (TDM), SIP (VoIP), IAX2 (VoIP) and a custom extension. By utilizing the various options, enclosed within each extension, it is possible to manipulate each extension s behavior. The custom extension can be utilized to program special functionality extensions, enabling the creation of extensions binded services, such as MicroBilling, Information gathering, etc. Once an extension had been defined, several optional parameters can be modified, in order to complement the default settings..special attention must be given to NAT traversal issues and DTMF issues, when working with IAX based extensions. You use a trunk to carry a call (or any number of calls) to a Voice Service Provider or a device that cares about what number you send to it (eg, another Asterisk /FreePBX Machine). There are 5 types of trunks supported: Zap Trunk Zap trunks provide connectivity to legacy TDM systems via Analog interfaces (FXO/FXS) or Digital interfaces (E1/T1). IAX2 Trunk IAX2 trunks provide interconnecting between Asterisk servers, utilizing the Inter-Asterisk Exchange Protocol. SIP Trunk SIP trunks provide interconnecting between Asterisk and SIP service providers, utilizing the Session Initiation Protocol. ENUM Trunk ENUM trunks utilize the e164.org number lookup services, and as a practice aren t used in generic PBX installations. Custom Trunk Custom trunks are available in order to configure any type of trunk which is not covered by the previous trunks, eg. H323, BRI ISDN, etc. Here in our implementation, though we have not Fig 12: Experimental set up of Asterisk based voice exchange 5. Conclusion and Future Work We expect that design and implementation presented in this paper will be a valuable developing guide for similar kind of operations. Asterisk based voice exchange provides us with a much better alternative solution. Its not only cost effective but also provides us with various features which we generally don t get with the conventional circuit switched based PBX. Moreover the system also provides for unlimited expansion and since it runs on a secure operating system like LINUX, its much less prone to viruses, worms and hackers. As far as future work is concerned, we would like to connect our Asterisk PBX with the conventional circuit switched networks with the help of PCI cards like for example Digium s TDM400P [7]. 6. References SERVER B [1] Andre du Toit, Private PBX Networks: Cost Effective Communication Solutions in Proc. IEEE 3 rd AFRICON Conference [2] Guo Fang Mao, Alex Talevski, Elizabeth Chang, Voice over Internet Protocol on mobile devices in Proc. 6 th IEEE ICIS [3] Md. Zaidul Alam, Saugata Bose, Md. Mhafuzur Rahman, Mohammad Abdullah Al-Mumin, Small office PBX using Voice over IP in Proc. IEEE ICACT 2007, Feb [4] Ryosuke Yamamoto, Fumikazu Iseki, Moo Wan Kim, Validation of VoIP System for University Network in Proc. IEEE ICACT 2008, Feb [5] Asterisk.org, "Features and Architecture of Asterisk PBX", accessed in March, [6] Taemoor Abbasi, Shekhar Prasad, Nabil Seddigh, Ioannis Lambadaris A comparative study of the SIP & IAX voice protocols in Proc. IEEE CCECE/CCGEI, Saskatoon, May 2005 [7] Anand Gorti A fault tolerant VoIP implementation based on open standards, in Proc. IEEE 6 th EDCC 06,