Making A Call with VoIP

Team name : VIP Project members: Camilo Devia and Gisoo Park

1. Abstract The Project we are proposing is to set up a VoIP system. To create a VoIP system, we first plan to use a program called Asterisk. This program will be run in virtual machines and will allows us to create a virtual server. The ultimate goal of doing this is to be able to use this server to set up an IP­PBX. Once Asterisk is properly set up, we plan to install softphones and/or install physical hardware to be able to make external calls. 2. Introduction VoIP is a concept both of us will like to explore, however we felt that it would be more appropriate to actually build a prototype for VoIP, due to the fact that we would gain more insight into this topic and be able to gain experience creating such a prototype. 2­1. IP­PBX IP­PBX can handle calls and connect between phones. PBX stands for Private Branch Exchange, which is a private telephone network used within a company. The users of the PBX phone system share a number of outside lines for making external phone calls. A PBX connects the internal telephones within a business and also connects them to the public switched telephone network (PSTN).One of the latest tendencies in PBX phone system development is the VoIP PBX, also known as IP PBX, which uses the Internet Protocol to transmit calls. IP PBX is a software­based PBX phone system solution which helps accomplish certain tasks and delivers services that can be difficult and costly to implement when using a traditional proprietary PABX. The Asterisk is one of the most well­known open source PBX. Asterisk is a software implementation of a hardware PBX and can run on a variety of hardware platforms. The features and benefits of owning an Asterisk PBX are numerous, and seemingly only limited by the imagination of the person who sets up and uses an Asterisk PBX. There are many ways of setting up an Asterisk PBX. A fully functional Asterisk PBX is to download and install a precompiled distribution such as Trixbox, PBX in a flash, Asterisk NOW, etc. Asterisk can be installed linux OSes(Debian, Ubuntu, Fedora, Mandrake, SuSE, etc). In this project we will cover the steps needed to install and set up the Trixbox Asterisk PBX. 2­2. SIP protocol SIP (Session Initiation Protocol) is based on RFC 2543 (Ref. 3) and is an application layer signaling protocol. It deals with interactive multimedia communication sessions between end users. It defines their initiation, modification and termination. SIP calls may be terminal­to­terminal, or they may require a server to intercede. If a server is to be involved, it is only required to locate the called party. For interworking with non­IP networks, Megaco and H.323 are required. Often, vendors of VoIP equipment integrate all three protocols on a single platform. SIP is closely related to IP. SIP borrows most of its syntax and semantics from the familiar HTTP (hypertext transfer protocol). A SIP message looks very much like an HTTP message, especially with message formatting, header and multipurpose Internet mail extension support. It uses addresses that are very similar to URLs and to email. For example, a call may be made to so­and­so@such­and­such. SIP messages are text­based rather than

binary. This makes writing easier and the debugging of software more straightforward. 2­3. Sip phones SIP Phones are the same thing as VoIP Phones or softphones. These are telephones that allow phone calls to be made using VoIP (voice over internet protocol) technology. There are two types of SIP Phones. The first type is the hardware SIP phone, which resembles the common telephone but can receive and make calls using the internet instead of the traditional PSTN system. SIP Phones can also be software­based. These allow any computer to be used as a telephone by means of a headset with a microphone and/or a sound card. A broadband connection and connection to a VOIP provider or a SIP server are also required. 2­4. VoIP Gateway A network device that converts voice and fax calls, in real time, between the public switched telephone network (PSTN) and an IP network. The primary functions of a VoIP gateway include voice and fax compression/decompression, packetization, call routing, and control signaling. Additional features may include interfaces to external controllers, such as Gatekeepers or Softswitches, billing systems, and network management systems. 2­5. IVR Interactive voice response (IVR) is a technology that allows a computer to interact with humans through the use of voice and DTMF tones input via keypad. In telecommunications, IVR allows customers to interact with a company’s host system via a telephone keypad or by speech recognition, after which they can service their own inquiries by following the IVR dialogue. IVR systems can respond with pre­recorded or dynamically generated audio to further direct users on how to proceed. IVR applications can be used to control almost any function where the interface can be broken down into a series of simple interactions. IVR systems deployed in the network are sized to handle large call volumes. IVR technology is also being introduced into automobile systems for hands­free operation. Current deployment in automobiles revolves around satellite navigation, audio and mobile phone systems. It's common in industries that have recently entered the telecommunications industry to refer to an automated attendant as an IVR. The terms, however, are distinct and mean different things to traditional telecommunications professionals, whereas emerging telephony and VoIP professionals often use the term IVR as a catch­all to signify any kind of telephony menu, even a basic automated attendant. 3. The Project Components 3­1 VMware Player ­ 5.0.0 build­812388 3­2. IP PBX Server ­ Trixbox 2.8.0.4

3­3. Sip phones 1) Softphone : X­Lite 5.0.0 2) Mobile Softphone : 3CXPhone 2.0.5 , Media5 fone 3.2 3) Sipphone : Cisco SPA535G2 3.4. SIP Gateway ­ Cisco SPA3102 4. Building IP­Telephony system 4­1. Install Trixbox IP ­PBX Step 1 : Download VMware and also Tribox From http://fonality.com/trixbox/

Step 2 : Mount Trixbox ISO using VMware Player.

Step 3 : Once Mounted, Run trixbox and when presented with this image press enter.

Step 4 : The setup will ask you for a username and password create one at your discretion. after this let the setup compress the packages. This should take 5­20 depending on you computer processing power.

Step 5 : Enter the username and password created before, and then type setup.

Step 6 : Since we are using the internet and not a local network put * on DHCP for

automatic mode. Press OK this will reboot the system.

Once It reboots the system will check the connections

Step 7 : Type ifconfig and record the IP address and you are done.

4­2. Setting Trixbox IP ­PBX 4­2­1. NAT setting (For connecting phones over the internet) Step 1 :After finished to install Trixbox, to connect web manager page using the IP. Default ID : maint , P.W : password

Step 2 :Click Config file editor

Step 3 :Find sip_nat.conf file. The file will be empty. Insert the following lines: Externip = your_external_ip_address localnet = internal.network.address_of_your machine/Subnet mask Then click on update and after restart the machine!

* Must open the ports in your firewall or router Ports

UDP Port 5060 is for SIP communication. UDP Port 5060­5082 range, SIP communications. TCP Port 5060 is for SIP but thought to be rarely used. UDP Port 10000 ­ 20000 is for RTP ­ the media stream, voice/video channel.

4­2­2. Making new stations Step 1 : Click PBX tab and click PBX settings tab

Step 2 : Click Extensions and Select Device : Generic SIP Device then hit submit button

Step 3:Enter User Extension ( ex : 1000), Must put numbers Enter Display Name

Step 4 ::Enter secret (which use password when a sipphone connect with IP­PBX)

Step 5 : Click Apply Configuration Change

Step 6 : Click Continue with reload

4­2­3. Making Trunk to connect a gateway for outbound call In this project, we used Cisco SPA­3102 gateway. The configuration of trunk is very various depend on gateways Step 1 : Click Trunks and Create SIP Trunk

Step 2 : Click Trunks and select add SIP Trunk Enter Outbound Caller ID : your pstn number(telephone number) Trunk Name : your trunk name

Step 3 : PEER Details ( This part is most important !!) Enter below comments in the PEER Details blank. disallow=all allow=ulaw canreinvite=no context=from-trunk dtmfmode=rfc2833 host=192.168.0.231 ; This IP address is gateway IP. Not Trixbox server IP. incominglimit=1 nat=never ; if the gateway is not on your local network you may need nat=yes port=5061 qualify=yes secret=1234 ; Write password you want. type=friend username=1-pstn ; Must match the trunk name

Step 4 : Click Submit Changes and remember to click the orange bar to update your system

4­2­4. Making Outbound Routes Step 1 : Click Outbound Routes Route Name : your route name whatever you want. Dial Patterns : 9441XXXX 91XXXXXXXXXX ­ The point here is to use dial patterns that will allow only those calls that you wish to go out via the gateway Trunk sequence: Select Trunk that we made above.

5. Setting SIP­phones 5­1. Setup soft sipphone (X­lite) Step 1 : Setting SIP Account

Account name : name User ID : Extension number Domain : IP­PBX Server IP or DNS Password : Extension password

5­2. Setup mobile sipphone (Midea 5 fone) * Can install both Apple and Android Step 1 : Install Media5 ­> Select Start button ­> Select Pre Configured List

Step 2 : Select Asterisk ­> Select Pre Configured List

Step 3 : Register the Trixbox Server and the Phone number. Title : name Username : Extension number Password : Extension password Address : IP­PBX Server IP or DNS

5­3. Setup Sipphone (Cisco SPA525G2) Step 1 : Checking the phone’s IP address for configuration Press the phone's setup key (Number 11) , Select Status ­> Network Status

Step 2 : Direct your browser to the SPA5xx's web user interface (web­UI) Enter advanced mode. (For example : http://192.168.0.25/admin/advanced)

Step 3 : Select Ext 1 tab.

Step 4 : Register the Trixbox server and the phone information. * Proxy and Registration ­ Proxy: mytrixbox.iptime.org:5060

­ Outbound Proxy: 67.22.26.2175060 * Subscriber Information ­ Display Name: 1006 ­ User ID: 1006 (phone number) ­ Password: 1006 (phone password)

Step 5 : Move Regional tab and Clear out the default vertical service activation codes.

Step 6 : Click Submit All Changes.

This will cause the phone to reboot. The phone's extension will register with the Trixbox server. Successful registration is indicated by the phone's extension LED burning green.

Step 7 : After finished registration , the phone will display five lines as same number.

Step 8 : You can modify this by changing the extension associated with each line key

in the Phone tab. Each extension can be associated with line 1 through N or the extension can be disabled if you do not want to use the line.

6. Conclusion: To create a VoIP system, we will first create a local IP PBX using Asterisk to create a local server that will allow us to connect hosts to it. This host could be both digital interfaces running on computers or physical phones. The ultimate goal is to eventually connect this local IP PBX to the outside world and be able to make calls to any line. The timeline for this ranges from september to “plan and prepare”, september­october( Setup) , october­ november (test), and november (implement). 7. References: Book 1) Goode, B.; , "Voice over Internet protocol (VoIP)," Proceedings of the IEEE , vol.90, no.9, pp. 1495­ 1517, Sep 2002 doi: 10.1109/JPROC.2002.802005 2) Upkar Varshney , Andy Snow , Matt McGivern , Christi Howard, Voice over IP, Communications of the ACM, v.45 n.1, p.89­96, January 2002 [doi>10.1145/502269.502271] 3)Khasnabish, B. (2003) Frontmatter, in Implementing Voice over IP, John Wiley & Sons, Inc., Hoboken, NJ, USA. doi: 10.1002/0471225274.fmatter

4) Meggelen, J., Smith, J., Madsen, L., Asterisk: The Future of Telephony. O'Reilly, 2007 Asterisk: The Future of Telephony. O'Reilly, 2007. O'Reilly, 2007 5) Gomillion, D., & Dempster, B. (2005). Building telephony systems with Asterisk an easy introduction to using and configuring Asterisk to build feature­rich telephony systems for small and medium businesses. Birmingham, U.K.: Packt Pub.. Internet 1) Ubuntu : How to install asterisk 10 on ubuntu 12.04 LTS « My Technical Notes. (n.d.). My Technical Notes. Retrieved September 10, 2012, from http://www.kartook.com/2012/05/ubuntu­how­to­install­asterisk­10­on­ubuntu­12­04­lts/ 2) default. (n.d.). Set Up Your Own IP­PBX. Charles Hayden's Home Page. Retrieved September 10, 2012, from http://chayden.net/Asterisk/SeUpAsteriskAtHome 3) Freeman, R. (2006, June 1). How does VoIP work? A technical guide to functional VoIP. Unified Communications information, news and tips ­ Search Unified Communications.com. Retrieved September 10, 2012, from http://searchunifiedcommunications.techtarget.com/feature/How­does­VoIP­work­A­technical­guide­to­functional­VoIP 4) Small Business PBX: The Basics ­ Cisco Systems . (n.d.). Cisco Systems, Inc. Retrieved September 10, 2012, from http://www.cisco.com/cisco/web/solutions 5)How Does VoIP Work? | voipreview.org. (2012, September 9). Voip ­ Voip Service ­ Voip Reviews | voipreview.org. Retrieved September 10, 2012, from http://www.voipreview.org/how_does_work Conference 1) Alam, M.Z.; Bose, S.; Rahman, M.M.; Abdullah Al­Mumin, M.; , "Small Office PBX Using Voice Over Internet Protocol (VOIP)," Advanced Communication Technology, The 9th International Conference on , vol.3, no., pp.1618­1622, 12­14 Feb. 2007 doi: 10.1109/ICACT.2007.358679 URL: http://ieeexplore.ieee.org/stamp/stamp.jsp?tp=&arnumber=4195481&isnumber=4195435 2) Kapicak, L.; Nevlud, P.; Zdralek, J.; Dubec, P.; Plucar, J.; , "Remote control of Asterisk via Web Services," Telecommunications and Signal Processing (TSP), 2011 34th International

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