1 VOICE OVER INTERNET PROTOCOL (VoIP): A VIEW AND A VISION KHALDOUN BATIHA Philadelphia University Head of Department of Business Networking and Systems Management Abstract The aim of this study is to explain the fundamentals of VoIP, regarding the functions and components and how it can be deployed by various organizations as an effective way of communication. In essence this study will try to answer the following questions: What does it mean to an organization to deploy VoIP? What makes up a VoIP solution and how can they take advantage of it?. Cisco router 9600 series and its peripheral devices such as VICs and Voice Network Modules in addition to the required router configuration software were used to verify the efficiency and reliability of VoIP over PSTN. The process used revealed that users will be able to send and receive calls as they would at same office or area, any where in the world, at no extra cost as call go via IP. This does not incur charges as call diversion does via PSTN, and the called party does not have to pay for the call. Once a general understanding of VoIP is achieved, organizations are better prepared to tackle the more complex issues that go into deploying a secure, low-cost, reliable and high-performance VoIP network. Keywords:- VoIP, PSTN, VICs, FXS, FXO, PBX, E&M, IP Telephony. INTRODUCTION Voice over IP ( VoIP ) is an important topic in information and communication technology circles, and increasingly within contact centers. From a world without VoIP just a few years ago, it is now becoming inevitable that VoIP will be deployed in everything from carriers backbones to our offices and homes. Indeed, leading industry analysts project that within the next four years traditional call center telephony infrastructure suppliers like Avaya, Alcatel, Nortel Networks and Siemens will announce their intention to discontinue support within five years for system architectures based on Time Division Multiplexing (TDM). The move from traditional circuit-switched to packet switched network technology will have a significant business impact on the applications used to record customer, supplier and other interactions, and the associated benefits such applications can deliver. The term Voice over Internet Protocol (VoIP) is used to refer to the practice of conveying live voice communication (including fax) via packet-switching technology utilizing Internet Protocol and transmitted over public or private data networks . This term thus includes the transmission of voice calls over the interconnected public Internet as well as the use of IP technology to convey voice calls over private leased point-to-point data networks . VoIP technology has the technological advantage that it is an easy to deploy technology, that, when coupled with readily available Internet connections, allows easy and cost effective carriage of voice communication over the Internet. In addition, VoIP technologies make use of packet-switching technology, which is more efficient in its use of network capacity than circuit switched technology. The result is that with VoIP voice traffic is carried more efficiently over data networks, which generally cost users much less to use than circuit switched network, this study will enable smaller businesses to access application functionality previously afforded only by large organizations, yet also affording larger customers greater flexibility in their infrastructure with resulting cost savings. IP Telephony is also enabling the development of enterprisewide contact management systems, which embrace every department that touches the customer, and every department that deals with the outside world. These external interactions with customers, suppliers and other third parties can include enormous amounts of valuable information and insight that, if made easily available to the right people in your business, may dramatically improve the service
2 that you offer to your customers, and the overall business performance. Until now, however, the valuable insight contained in these telephone calls has been, at best, only captured and shared within the contact center. In this study I have showed the possibilities of deploying VoIP technology in Jordanian private and public sectors by covering the fundamentals and advance need to configure this technology in small business. This has been developed using Cisco router and other material as hardware equipments and Cisco router IOS Software commands for necessary configurations, this study as well contain a real practical example shows in reality the overall architecture of VoIP technology, and real implementations and configurations needed to deploy this technology, and shows it's interaction with public switching telephone network (PSTN), in addition to solutions to challenges of deploying this technology it real telecommunication systems for more effective and smoothly use. VoIP IMPLEMENTATION CHALLENGES Because IP does not by default provide any mechanism to ensure that data packets are delivered in sequential order, or provide any Quality of Service guarantees, implementations of VoIP face problems dealing with latency and possible data integrity problems.one of the central challenges for VoIP implementers is restructuring streams of received IP packets, which can come in any order and have packets missing, to ensure that the ensuing audio stream maintains a proper time consistency. Another important challenge is keeping packet latency down to acceptable levels, so that users do not experience significant lag time between when they speak and the signal is decoded on the other end of the connection. Solutions to these problems include: Certain hardware solutions can distinguish VoIP packets and provide priority queuing for this class of service. Alternatively packets can be buffered but this can lead to an overall delay similar to that encountered on satellite circuits. The network operator can also ensure that there is enough bandwidth end-to-end to guarantee lowlatency low-loss traffic: this is easy to do in private networks, but much harder to do in the public Internet. Jitter (delay variance) problems are mainly generated in lowband access (less than 256 kbit/s) because of serialization of big (1500 bytes) data packets. At these rates, fragmentation mechanisms for these big packets are needed (interleaving small voice packets) to reduce the delay. Over networks slower than 256 kbit/s it is almost impossible to ensure quality voice without a fragmentation mechanism. HARDWARE CONFIGURATION The two types of hardware of VoIP feature for Cisco 3600 series routers uses: Voice network modules Voice interface cards (VICs) The configuration of these types of hardware should be taken (voice network module and VIC) before performing software configuration tasks. Voice network modules install in a slot in a Cisco 3600 series router, and convert telephone voice signals into a form that can be transmitted over an IP network. The one-slot voice network module provides one slot for a voice interface card. The two-slot voice network module provides two slots for voice interface cards. Voice interface cards install in slots in the voice network module, and provide connections to the telephone equipment or network. The VICs interfaces include: Foreign Exchange Station (FXS), this interface connects directly to a standard telephone. Foreign Exchange Office (FXO), this interface connects local call to a public switching telephone network, or to Private Branch Exchange (PBX). Ear and Mouth (E&M), signaling technique for two-wire and four-wire telephone and trunk interfaces. The E&M VIC connects remote calls from an IP network to a PBX for local distribution. To connect VICs to network, I used a standard Rj-11 modular telephone cable to connect FXS VIC ports to telephone or fax then I used Rj-11 modular telephone cable to connect FXO VIC ports to the PSTN or PBX, through a telephone wall outlet, and the E&M VIC uses RJ-48S connector and cable.
3 IOS SOFTWARE CONFIGURATION MODE The main thing to be considered before entering to software configuration mode is the voice port numbering, to display the available ports numbering after finishing hardware configuration, is to use IOS show voice port command. To be able to use IOS configuration mode, first connects the console cable to the router then power up the router, after a few seconds, the EXEC prompt will be display. Type enables then types the password to enter to enable mode Router> enable Password: As being familiar to router configuration, the configuration change can be made only by using enable mode. To enter to configuration mode Router# config terminal Router(config)# If the router is not being configured before, the routing protocols should be configured as a follow Router(config)# ip routing Router(config)# appletalk routing Router(config)# ipx routing These configurations are general to all router configurations, and should be performed before proceeding in further configurations. In this time the Cisco 9600 router will be ready for VoIP necessary configuration. FXS INTERFACES CONFIRGURATION As I have mentioned before, the ports on FXS VICs connected directly to a standard telephone, fax machine, or similar device. Before configuring FXS interfaces, and to make this section and other sections easy to understand, I have used the Philadelphia 1 as router hostname, and other routers will be naming sequentially Philadelphia 2, Philadelphia 3 etc. Table 1: Philadelphia 1 Router Telephone Numbers and Voice Ports Telephone Number Voice Port /0/ /0/ Figure (1) Basic Voice Network (Philadelphia 1) additional telephones and fax machine could be connected in this example, up 12 in total in case of using Cisco 3640 router, the remaining voice network module provide interfaces for IP connectivity to the LAN or WAN and for data traffic, and for more than 12 devices, we cam add more routers, or to use an E&M VIC and a local PBX, rather than connecting every telephone to its own FXS VIC. To
4 make routing a received voice call to the right destination, the router needs to know which telephone number belongs to each voice port. In other words, router needs to know the information in table 1. To hold the information on the table 1, Cisco IOS software uses objects called dial peers. the information including voice port, telephone numbers, and other parameters for call are put together associating them all with the same dial peer. Dial peers identified by numbers, but to avoid confusing these numbers with telephone numbers, they are usually referred to as a tags. Dial peers tags are integers that can range from 1 to ( ) Dial peers on the same router must have unique tags, but we can use reuse the tags on other routers. Table 2 assigns each telephone number-voice port pair on the Philadelphia 1 router a dial peer tag. Within the allowed range, we can choose any dial peer tag or any system that is convenient. This type of dial peer is called a plain old telephone service (pots) dial peer or local dial peer means that the dial peers associates a physical voice port with a local telephone device. Table 2: Philadelphia 1 Router Local Dial Peers Telephone number Voice Port Dial Peer Tag /0/ /0/1 213 Now using the global configuration mode, I configure the router with the information in above table. Philadelphia1 (config)# dial-peer voice 211 pots Philadelphia1 (config-dial-peer)# dest-pat Philadelphia1 (config-dial-peer)# port 2/0/0 Philadelphia1 (config)# dial-peer voice 213 pots Philadelphia1 (config-dial-peer)# dest-pat Philadelphia1 (config-dial-peer)# port 2/0/1 Philadelphia1 (config-dial-peer)# exit Philadelphia1 (config)# Cisco IOS software refers to a telephone numbers as a destination pattern, because it is the destination pattern for an incoming or outgoing call. A destination pattern always begins with (+) signs. It can also include asterisks (*) and pound signs (#) from the telephone keypad, and commas (,) and periods (.), which have special meaning. Parentheses (()), hyphens (-), slashes (/), and spaces ( ), which are often used to make telephone numbers easier for humans to read, are not allowed. I supposed that Philadelphia has launched a new branch in south Jordan. Figure(2) shows the Philadelphia 2 network, and table 3 list phone numbers, voice pots, and dial peer tags for this branch.
5 Figure (2) Basic Voice Network (philadelphia2) Table 3 Philadelphia 2 Router Local Dial Peers Telephone Number Destination Pattern Voice Port Dial Peer Tag /1/ /1/1 802 Philadelphia2 (config)# dial-peer voice 801 pots Philadelphia2 (config-dial-peer)# dest-pat Philadelphia2 (config-dial-peer)# port 3/1/0 Philadelphia2 (config)# dial-peer voice 802 pots Philadelphia2 (config-dial-peer)# dest-pat Philadelphia2 (config-dial-peer)# port 3/1/1 Philadelphia2 (config-dial-peer)# exit Philadelphia2 (config)# In office PBXs are usually configured so a user can dial a local call (within the same PBX) by dialing the extension only for instance, the four-digit extension 3425 or the five-digit extension instead of the full telephone number, I provide the same shortcut on a voice over IP network by using the number-expansion (num-exp) command. This command tells the router to expand a particular sequence of dialed numbers into a complete telephone number (destination pattern). For instance, to expand 3425 into , enter the following command: Philadelphia2 (config)# num-exp To expand 4141 into , enter the following command: Philadelphia2 (config)# num-exp More generally, you can use the period (.) as a wild-card character representing a single digit. For instance, the command Pheladelphia2 (config)# num-exp Expands any dialed sequence of four digits by prefixing to it. To use five-digit extensions beginning with the numeral 5 rather than four-digit extensions, you would enter the following command: Philedlphia2 (config)# num-exp The corresponding commands for the East router would be (for four-digit extensions): Philadelphia2 (config)# num-exp Or for five-digit extensions: Philadelphia2 (config)# num-exp Now it is possible to place calls between telephones connected to same router using extension instead of using the full telephone number. Naturally, the both branches would like to send voice traffic to each other over the same IP network they use for data traffic. A WAN port of some type on each router connects to the IP WAN. As we know the IP routers know how to locate IP addresses on the network, but they do not know how to
6 locate telephone numbers. to route an outgoing voice call over routers connection. The Philadelphia2 router has to associate a telephone number in the Philadelphia1 with the IP address of the Philadelphia1 router. This is done by associating both pieces of information with a remote dial peer or VoIP dial peer on the philadelphia2 router as shown in table 4: Table 4: philadelphia2 Router Remote Dial Peers Remote Telephone Destination IP Address Dial Peer Location Number Pattern Tag Philadelphia Philadelphia It is easier to use periods as a wildcard as in the following table: Table 5: philadelphia2 Router Remote Dial Peers with Wild Cards Remote Telephone Destination IP Address Dial Peer Location Number Pattern Tag Philadelphia **** Now enter to the philadelphia2 global configuration mode and enter the following commands: Philadelphia2 (config)# dial-peer voice 601 voip Philadelphia2 (config-dial-peer)# dest-pat Philadelphia2 (config-dial-peer)# session-target ipv4: In Cisco IOS software the remote network is known as session target. In this example this command followed by the IP address of the remote router. We can use the prefix dns followed by DNS name, for instance: Philadelphia2 (config-dial-peer)# session-target dns: voice.philadelphia1router.com The thing can be easier by configuring number expansion for philadelphia1 telephone numbers on the philadelphia2 router: Philadelphia2 (config)# num-exp Now the user in philadelphia2 branch can dial a five-digit extension beginning with 8 to reach a telephone on the philadelphia1 area. Now the philadelphia2 router is configured to send calls to the philadelphia1 router, this is shown in table 6: Table 6 Philadelphia1 Router Remote Dial Peers with Wild Cards Remote Location Telephone Number IP Address Dial Peer Tag Philadelphia **** Here it is good to configure RSVP on the WAN interface, and must configure each VOIP dial peer to request an RSVP session, to configure RSVP on WAN interface I used following commands: Philadelphia1> enable Password: Philadelphia1# configure terminal Philadelphia1 (config)# interface serial 0/0 Philadelphia1 (config-if)# ip rsvp bandwidth Philadelphia2> enable Password: Philadelphia2# configure terminal Philadelphia2 (config)# interface serial 0/0
7 Philadelphia2 (config-if)# ip rsvp bandwidth Philadelphia1 (config-dial-peer)# req-qos controlled-load Philadelphia2 (config-dial-peer)# req-qos controlled-load Otherwise no bandwidth is reserved for voice traffic. If the bath between endpoints of a voice call travels through intermediate routers, and these routers should configure for local and remote dial peers if they have voice devices attached. FXO INTERFACES CONFIRGURATION The main function of FXO interfaces provide a gateway from the VOIP network to the analog PSTN, or to a PBX that does not support E&M signaling. So users can reach telephones and fax machines outside the VOIP network. To create a local dial peer for an FXO interface, the destination pattern refers to outgoing calls, and including wild card in it, because the PSTN performs the switching. The VOIP feature can also remove digits that we don't want to send to the PSTN, I supposed the user want to dial 7 to reach an outside line (analog PSTN), the configuration can be made as a following: Philadelphia2 (config)# dial-peer voice 201 pots Philadelphia2 (config-dial-peer)# dest-pat Philadelphia2 (config-dial-peer)# port 1/0/0 To enable philadelphia1 router users to make calls over the philadelphia2 router's local PSTN, the following commands is used: Philadelphia1 (config)# dial-peer voice 701 voip Philadelphia1(config-dial-peer)# dest-pat Philadelphia1(config-dial-peer)# session-target ipv4: Philadelphia2(config)# dial-peer voice 601 pots Philadelphia2(config-dial-peer)# dest-pat Philadelphia2(config-dial-peer)# port 1/0/0 The philadelphia1 router now sends all calls whose numbers begin with special prefix 8 over the ip network to the philadelphia2 router. the philadelphia2 router removes the 8 and passes the calls through its analog FXO gateway to the local PSTN. E&M INTERFACES CONFIGURATION If we have more than a few voice users at each location, then the cost of voice ports and routers, and the effort needed to configure dial peers for all the combinations of origins and destinations, increases rapidly. In this situation, it may be more efficient to use a PBX at each location to switch local traffic and to direct incoming calls, and to connect the PBXs over an IP network using E&M voice interface cards. Consider the Philadelphia braches, now each branch has a PBX to operate its internal telephone network. While the IP network carries voice traffic between the branches. In the following figure, each BPX connects to the IP router over an E&M interface connection. Figure (3) Linking PBXs over the IP Network (Local Dial Peers) **** E&M VIC IP Cloud **** 3/0/0 DESTINATION SOURCE I supposed both BPX use E&M interface type 2, with four-wire operation and immediate-start signaling. The value used in coming configuration depends on PBX, and provided from the concern telecommunication department or the PBX manufacturer.
8 In the following configuration, philadelphia2 users can dial 8 and a four-digit extension to reach telephones in the philadelphia1 branch. Philadelphia1 users can dial 5 and a four-digit extension to reach telephones in the philadelphia2 branch The philadelphia2 router is connected to the PBX over E&M port 3/0/0. That means that this port is associated with local (POTS) dial peers for incoming calls. But we no longer need to associate every telephone number with its own port. Instead, I can configure a local dial peer as if all the philadelphia2 telephones (represented by a wild-card destination pattern) are connected directly to this port, as shown in the following commands: Philadelphia2 (config)# dial-peer voice 111 pots Philadelphia2 (config-dial-peer)# dest-pat Philadelphia2 (config-dial-peer)# port 3/0/0 Remote (VoIP) dial peers for outgoing calls associate destination phone numbers on the philadelphia1 router with that router s IP address: Philadelphia2 (config)# dial-peer voice 121 voip Philadelphia2 (config-dial-peer)# dest-pat Philadelphia2 (config-dial-peer)# session-target ipv4: Philadelphia2 (config-dial-peer)# exit Philadelphia2 (config)# Philadelphia2 (config)# num-exp To configure the E&M port similarly to any other network interface, using the follwing command: Philadelphia2 (config)# voice-port 3/0/0 Philadelphia2 (config-voice-port)# signal immediate Philadelphia2 (config-voice-port)# operation 4-wire Philadelphia2 (config-voice-port)# type 2 To configure philadelphia1 router, the PBX for this router is connected to E&M port 2/0/1. the following commands configure a local dial peer for all philadelphia1 telephones: Philadelphia1 (config)# dial-peer voice 211 pots Philadelphia1 (config-dial-peer)# dest-pat Philadelphia1 (config-dial-peer)# port 2/0/1 The following commands configure dial peer for telephones on the philadelphia2 router: Philadelphia1 (config)# dial-peer voice 221 voip Philadelphia1 (config-dial-peer)# dest-pat Philadelphia1 (config-dial-peer)# session-target ipv4: Philadelphia1 (config-dial-peer)# exit Philadelphia1 (config)# To configure number expansion, to make it easy for philadelphia1 users to dial numbers on the philadelphia2 router: Philadelphia2 (config)# num-exp To configure the E&M port: Philadelphia1 (config)# voice-port 2/0/1 Philadelphia1 (config-voice-port)# signal immediate Philadelphia1 (config-voice-port)# operation 4-wire Philadelphia1 (config-voice-port)# type And to save all the configuration, to each router the following commands is used: Press Ctrl-Z. Router# show running-config Router# copy running-config startup-config Building configuration... [OK] Router# now the router is configured to boot in the new configuration.
9 CONCLUSION This study showed that call communications system providers would be able to easy install and deploy the VoIP technology where ever they need. As stated before, this technology will assist various types of organizations to provide more reliable, efficient, upgradeable, low-cost and secure way of communication. Although few office environments and even fewer homes use a pure VoIP infrastructure, telecommunications providers routinely use IP telephony, often over a dedicated IP network, to connect between their switching stations, where they convert the dedicated voice signal to IP packets and back. The result is a data-abstracted digital network which the provider can easily upgrade and use for multiple purposes. Corporate customer support centers which provide support over telephone often use IP telephony exclusively to take advantage of the data abstraction that comes with it. The benefit of using this technology is the need for only one class of circuit connection and better use of the available bandwidth. IP telephony is commonly used to route traffic that may be originated from and terminated at conventional PSTN telephones. VoIP is now widely deployed by carriers, especially for international telephone calls. Most commonly, users are completely unaware that their telephone call is being routed over IP infrastructure for most of its distance, instead of entirely over the circuit switched PSTN. This study will lead to the use of VoIP also by large companies to eliminate call charges between their offices, by using their data network to carry inter-office calls. They may also use VoIP to reduce the costs of calls outside the company, by carrying them to the nearest point on their network before handing them off to the PSTN by offering a gateway to the PSTN from any VoIP phone. REFERANCES . ETSI Guide , Speech processing, Transmission and Quality aspects (STQ); Overall transmission plan aspects for telephony in a private network, November . D. De Vleeschauwer, J. Janssen, G. H. Petit, F. Poppe, Quality Bounds for Packetized Voice Transport, Alcatel Telecom Review, First quarter 2000, pp , January . D. De Vleeschauwer, A. Van Moffaert, M.J.C. Büchli, J. Janssen, G.H. Petit, Quality Issues for Packet-Based Voice Transport, Telektronikk 2/ Internet Traffic Engineering, pp , September . Bagain. H," Jordan Communication Projections Report", Arab Advisors Group Strategic Research Service Communications, September  Marjory S. Blumenthal, David D. Clark, Rethinking the design of the Internet: the end-to-end arguments vs. the brave new world, ACM Transactions on Internet Technology (TOIT), v.1 n.1, p , Aug  Bond, G. W Fault management issues for a next-generation IP telecom services architecture. In Workshops and Abstracts of the International Conference on Dependable Systems and Networks (DSN 2000), pages D-11--D-13, June.  Bond, G. W. and Goguen, H ECharts: Balancing design and implementation. In Proceedings of the 6th IASTED International Conference on Software Engineering and Applications, pages ACTA Press.  Bond, G. W., Ivančić, F., Klarlund, N., and Trefler, R ECLIPSE feature logic analysis. In Proceedings of the Second IP Telephony Workshop, pages Columbia University, New York, NY, April.
10  H. Velthuijsen, L. G. Bouma, Feature Interactions in Telecommunications Systems, IOS Press, Amsterdam, The Netherlands, 1994  M. Calder, E. Magill, Feature Interactions in Telecommunications and Software Systems VI, IOS Press, Amsterdam, The Netherlands, 2000  Cameron, E. J., Griffeth, N. D., Lin, Y.-J., Nilson, M. E., Schnure, W. K., and Velthuijsen, H A feature-interaction benchmark for IN and beyond. IEEE Comm. 31, 3, , March.  K. E. Cheng, T. Ohta, Feature Interactions in Telecommunications III, IOS Press, Amsterdam, The Netherlands, 1995  Cheung, E., Jackson, M., and Zave, P Distributed media control for multimedia communications services. In Proceedings of the 2002 IEEE International Conference on Communications, Symposium on Multimedia and VoIP---Services and Technology, IEEE Communications Society, 02CH37333C.  Choi, S., Turner, J., and Wolf, T Configuring sessions in programmable networks. In Proceedings of IEEE Infocom.  D. Clark, The design philosophy of the DARPA internet protocols, ACM SIGCOMM Computer Communication Review, v.18 n.4, p , August 1988  P. Drini, L. Logrippo, R. Boutaba, Feature Interactions in Telecommunications Networks IV, IOS Press, Amsterdam, The Netherlands, 1997  Duran, J. M. and Visser, J International standards for intelligent networks. IEEE Communications 30, 2, , February.  Emden R. Gansner, John M. Mocenigo, Stephen C. North, Visualizing software for telecommunication services, Proceedings of the 2003 ACM symposium on Software visualization, June 11-13, 2003, San Diego, California  Emden R. Gansner, Stephen C. North, An open graph visualization system and its applications to software engineering, Software Practice & Experience, v.30 n.11, p , Sept  Garrahan, J. J., Russo, P. A., Kitami, K., and Kung, R Intelligent Network overview. IEEE Communications 31, 3, , March.  Hall, R. J Feature interactions in electronic mail. In Feature Interactions in Telecommunications and Software Systems, M. Calder and E. Magill, Eds. pages  Michael Jackson, Pamela Zave, Distributed Feature Composition: A Virtual Architecture for Telecommunications Services, IEEE Transactions on Software Engineering, v.24 n.10, p , October 1998  Jackson, M. and Zave, P The DFC Manual. AT&T Research Technical Report, August. Available at  URL: -  Kristensen, A. SIP Servlet API, Version 1.0. Dynamicsoft, Inc.  McGlashan, S., Burnett, D., Danielsen, P., Ferrans, J., Hunt, A., Karam, G., Ladd, D., Lucas, B., Rehor, K., Porter, B., and Tryphonas, S VoiceXML 2.0, W3C Working Draft, 24 April.
11  Stephen C. North, Gordon Woodhull, Online Hierarchical Graph Drawing, Revised Papers from the 9th International Symposium on Graph Drawing, p , September 23-26, 2001  Bhaskaran Raman, Helen J. Wang, Jimmy Shih, Anthony D. Joseph, Randy H. Katz, The Iceberg Project: Defining the IP and Telecom Intersection, IT Professional, v.1 n.6, K. Kimbler, L. G. Bouma, Feature Interactions in Telecommunications and Software Systems V, IOS Press, Amsterdam, The Netherlands, 1998 APPENDIX A. Rules and Regulations Regarding the Implementation of VoIP Technology The statement is to be issued by the TRC in order to bring clarity and certainty to the regulatory treatment of Voice over Internet Protocol (VoIP) technologies. This issue has been complex for operators, service providers and users in Jordan, as VoIP technology tends to blur the lines between voice service and data service. This is problematic in part because the regulatory regime in Jordan, in particular the License of Jordan Telecommunications Company (Jordan Telecom) makes a clear distinction between voice and data services and treats them differently as a legal and regulatory matter. This statement does not attempt to define voice or data any more specifically than it already is defined, but rather attempts to clarify that the existing rules govern the activities of service providers and users differently. The TRC has undertaken an extensive review of this issue, including consideration of the public consultation conducted by the Ministry of Information and Communications Technology (MoICT), as well as the treatment of this issue by telecommunications regulators in other jurisdictions. The Statement s primary concern is the scope of the exclusivity currently enjoyed by Jordan Telecom for the provision of Public Switched Voice Service. This exclusivity is contained in its License and expires December 31, This statement was released in draft form on August 14 th and comments were invited from interested parties. Comments were received from BossIT, TEData, Jordan Telecom, Cyberia, Batelco Jordan and Khaled Hudhud. The TRC has taken all comments received into account in formulating its response .
12 B. FORMAL STATEMENT REGARDING VOIP The TRC views the duty to foster the development of innovative telecommunications technologies in Jordan to be at the core of the strategic goals of the TRC and of Jordan. At the same time, the TRC is bound by the terms and conditions of the Telecommunications Law and the licenses and other regulatory instruments it has issued. It takes seriously its duty to ensure regulatory certainty by making its rulings as clear and consistent with one another and the Telecommunications Law as possible. Balancing these two duties is one of the TRC s most important challenges. The TRC therefore issues this formal statement to clarify the application of Jordan Telecommunications Company s license terms to the use of Voice over Internet Protocol technologies in Jordan. 1. The TRC views voice service provided using VoIP technology as the functional equivalent of voice service provided using circuit switched voice technology. Jordan Telecom therefore possesses the exclusive right to provide Public Switched Voice Service using VoIP technologies pursuant to Section 2.4 of its License. 2. Jordan Telecom s exclusive right to provide Public Switched Voice Service covers only the provision of a commercial service by one party to another separate party. Persons in Jordan are therefore free to utilize their legitimately subscribed data communication circuits to transmit and receive voice or other media format of their choosing without fear of infringing on Jordan Telecom s License, as long as they are transmitting and receiving such information for their own private use and not on behalf of a third party. 3. The use of VoIP technologies by commercial service providers other than Jordan Telecom to provide voice service to the public is prohibited prior to January 1, After that time, the provision of voice service using VoIP will be governed by future rules and/or guidelines to be issued by the TRC. The current prohibition includes services that utilize a normal telephone connected to the public switched telephone network, which then routes calls over the Internet via a gateway operated by a service provider, as well as the commercial offering of voice service to the public by the owner of a personal computer or other device connected to the Internet. 4. No entity other than Jordan Telecom may offer voice service to the public utilizing VoIP prior to January 1, The TRC views any activity by data licensees to market VoIP services to its users as equivalent to offering Public Switched Voice Services and therefore prohibited by Jordan Telecom s exclusive right to provide such services. The TRC therefore prohibits any activity by Data Communication Licensees to market voice service via VoIP to the public in Jordan using prepaid cards, hardware devices with included payment functionality or software.
13 5. The use of VoIP technologies prior to January 1, 2005 by commercial service providers other than Jordan Telecom to convey foreign-originated voice calls for termination on the public switched telephone network in Jordan, including on the network of mobile operators, constitutes an infringement of Jordan Telecom s License. The use of such technologies after December 31, 2004 will be subject to applicable TRC decisions to be adopted n the future. 6. Any infringement of Jordan Telecom s license by any entity providing commercial voice service using VoIP should be brought to the immediate attention of the TRC, which will take appropriate measures. The TRC will also take active measures to detect such activity and prosecute offending parties. 7. No Internet service provider, including Jordan Telecom, is authorized to block or otherwise interfere with the activities of any Internet user in Jordan in accessing or communicating with users or other entities outside of Jordan using VoIP without express prior authorization from the TRC. 8. Any person who provides voice service to the public utilizing VoIP in contravention of this decision or of the License of Jordan Telecom will be subject to the sanctions of the law including Articles and of the Telecommunications Law. Such enforcement action may result in seizure of equipment, financial penalties, prison terms or a combination of the above. The TRC will endeavor to apply the strictest penalties available in the event of a violation of these provisions.
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CVOICE Exam Topics Cisco Voice over IP Exam #642-432 7/14/2005 From the Cisco CVOICE 642-432 Exam Topics Voice over IP Technologies Describe the similarities and differences between PSTN and VoIP including
Course Title: Prerequisites: Training Program (5 months) IP Implementation in Private Branch Exchanges Must fresh graduates Communication/Electronics Engineers" 1- Soft Skills Training (4 weeks) 1. Communication
Lesson 1 Abstract Introduction to VoIP Technology 2012. 01. 06. This first lesson of contains the basic knowledge about the terms and processes concerning the Voice over IP technology. The main goal of
Call Setup and Digit Manipulation End-to-End Calls This topic explains how routers interpret call legs to establish end-to-end calls. End-to-End Calls IP Telephony 2005 Cisco Systems, Inc. All rights reserved.
Chapter 3 - Voice Dial Peers In order to understand the concept of dial peers, it is important to understand call legs. A voice call over a packet network is segmented into discrete call legs. A call leg
Cisco Voice Gateways PacNOG6 VoIP Workshop Nadi, Fiji. November 2009 Jonny Martin - firstname.lastname@example.org Voice Gateways Any device with one or more TDM PSTN interfaces on them TDM - Time Division Multiplexing
Integration of Voice over Internet Protocol Experiment in Computer Engineering Technology Curriculum V. Rajaravivarma and Farid Farahmand Computer Electronics and Graphics Technology School of Technology,
Frequently Asked Questions about Integrated Access Phone Service How are local, long distance, and international calls defined? Local access transport areas (LATAs) are geographical boundaries set by the
KEY VOIP TERMS 1 ACD: Automatic Call Distribution is a system used to determine how incoming calls are routed. When the ACD system receives an incoming call it follows user-defined specifications as to
Voice over IP (VoIP) Basics for IT Technicians VoIP brings a new environment to the network technician that requires expanded knowledge and tools to deploy and troubleshoot IP phones. This paper provides
52-30-20 DATA COMMUNICATIONS MANAGEMENT INTRODUCTION TO VOICE OVER IP Gilbert Held INSIDE Equipment Utilization; VoIP Gateway; Router with Voice Modules; IP Gateway; Latency; Delay Components; Encoding;
Mapping Outbound Calls to Unique FXS/FXO Ports on Analog Gateways Document ID: 41887 Contents Introduction Prerequisites Requirements Components Used Conventions Analog Gateways Gateway Protocol Support
Integrating VoIP Phones and IP PBX s with VidyoGateway Updated February 2011 INDEX: I. ABSTRACT.1 II. III. IV. VIDYOGATEWAY OVERVIEW.. 1 NETWORK TOPOLOGIES AND DEFINITIONS...2 CONNECTING TO VIDYOCONFERENCES
Building a Scalable Numbering Plan Scalable Numbering Plan This topic describes the need for a scalable numbering plan in a VoIP network. Dial Plans Dial plans contain specific dialing patterns for a user
Internet Telephony Terminology Understanding the business phone system world can be a daunting task to a lay person who just wants a system that serves his or her business needs. The purpose of this paper
Methods for Lawful Interception in IP Telephony Networks Based on H.323 Andro Milanović, Siniša Srbljić, Ivo Ražnjević*, Darryl Sladden*, Ivan Matošević, and Daniel Skrobo School of Electrical Engineering
IP Telephony Basics Part of The Technology Overview Series for Small and Medium Businesses What is IP Telephony? IP Telephony uses the Internet Protocol (IP) to transmit voice or FAX traffic over a public
WAN Data Link Protocols In addition to Physical layer devices, WANs require Data Link layer protocols to establish the link across the communication line from the sending to the receiving device. 1 Data
FACILITY TELECOMMUNICATIONS MANAGEMENT FOR THE GOVERNMENT EMERGENCY TELECOMMUNICATIONS SERVICE Introduction This document provides telecommunications management information for organizations that use the
Network Connection Considerations for Microsoft Response Point 1.0 Service Pack 2 Updated: February 2009 Microsoft Response Point is a small-business phone solution that is designed to be easy to use and
Unified Messaging with Microsoft Exchange Server 2007 Dell Product Group Dell White Paper BY FARRUKH NOMAN SUMANKUMAR SINGH September 2007 CONTENTS EXECUTIVE SUMMARY...3 INTRODUCTION...4 EXCHANGE UNIFIED
TAXONOMY OF TELECOM TERMS Prepared by TUFF Ltd This short taxonomy is designed to describe the various terms used in today s telecommunications industry. It is not intended to be all embracing but to describe
IP- PBX Functionality Options With the powerful features integrated in the AtomOS system from AtomAmpd, installing & configuring a cost- effective and extensible VoIP solution is easily possible. 4/26/10
Network Scenarios Pagina 1 di 35 Table of Contents Network Scenarios Cisco 827 s Network Connections Internet Access Scenarios Before You Configure Your Internet Access Network Replacing a Bridge or Modem
ehealth and VoIP Overview Voice over IP (VoIP) configurations can be very complex. Your network could contain a variety of devices, applications, and configuration capabilities to support voice traffic.
Agilent Technologies Performing Pre-VoIP Network Assessments Application Note 1402 Issues with VoIP Network Performance Voice is more than just an IP network application. It is a fundamental business and
Article VoIP Introduction Internet telephony refers to communications services voice, fax, SMS, and/or voice-messaging applications that are transported via the internet, rather than the public switched
VoIP H.323 Series VoIP Gatways: VoIP 422/404/440/800 VoIP Routers: VoIP 404R/440R/200R/110R Quick Setup Guide Important Information In this guide, you will learn how to setup our VoIP gateway and Routers
Traditional PBX & Hosted VOIP Technology: The Key Differences & What They Mean For Your Business CONTENTS Summary... 3 What s The Buzz About?... 3 What It Means For Businesses... 3 What It Means For Employees...
Configuring Network Side ISDN PRI Signaling, Trunking, and Switching This chapter describes the Network Side ISDN PRI Signaling, Trunking, and Switching feature. The following main sections are provided:
Cisco Voice Gateways PacNOG 3 VoIP Workshop June 2007, Cook Islands Jonny Martin email@example.com Voice Gateways Any device with one or more TDM PSTN interfaces on them TDM - Time Division Multiplexing
International Journal of Information & Computation Technology. ISSN 0974-2239 Volume 4, Number 14 (2014), pp. 1373-1378 International Research Publications House http://www. irphouse.com Call Routing through
1 Voice over IP (VoIP) - Basic Voice over IP - Voice over IP (VoIP) is a digital form of transport for voice transmissions, replacing analog phone systems. The benefits of VoIP are considerable: Better
Table of Contents Mapping Outbound VoIP Calls to Specific Digital Voice Ports...1 Introduction...1 Before You Begin...1 Conventions...1 Prerequisites...1 Components Used...1 Configure...2 Network Diagram...2
IP Telephony v1.0 Scope and Sequence Cisco Networking Academy Program Table of Content COURSE OVERVIEW...4 Course Description...4 Course Objectives...4 Target Audience...5 Prerequisites...5 Lab Requirements...5
Table of Contents Device SIP Trunking Administrator Manual Version 20090401 Table of Contents... 1 Your SIP Trunking Service... 2 Terminology and Definitions... 2 PBX, IP-PBX or Key System... 2 Multi-port
Implementing Cisco IOS Telephony and Unified Communications Express (IITUCX) Course Objectives Explain the benefits and components of a Cisco Unified Communications system Describe how traditional telephony
Session Title: Exploring Packet Tracer v5.3 IP Telephony & CME Scenario With the scheduled release of Packet Tracer v5.3 in the near future, this case study is designed to provide you with an insight into
How Small Businesses Can Use Voice over Internet Protocol (VoIP) Internet Technology for Voice Communications Small businesses will find this booklet useful for learning how VoIP works and for clarifying
BUILDING LARGE CAMPUS ASTERISK-BASED PABX SYSTEMS.0 Background The abundance of Local Area Networks such as those found on most African Universities, and recent advances in Computer and Telephony integration
Requirements of Voice in an IP Internetwork Real-Time Voice in a Best-Effort IP Internetwork This topic lists problems associated with implementation of real-time voice traffic in a best-effort IP internetwork.
White paper An introduction Table of contents 1 Introducing 3 2 How does it work? 3 3 Inside a normal call 4 4 DTMF sending commands in sip calls 6 5 Complex environments and higher security 6 6 Summary
CHAPTER 1 Background Traditional PSTN Equipment Traditional telephone services are engineered and offered over the public switched telephone network (PSTN) via plain old telephone service (POTS) equipment
Using Analog E&M Ports to Interface to Overhead Paging Systems Document ID: 27627 Contents Introduction Prerequisites Requirements Components Used Conventions Background Information Configure Network Diagram
WAN Technology Heng Sovannarith firstname.lastname@example.org Introduction A WAN is a data communications network that covers a relatively broad geographic area and often uses transmission facilities provided
Future of Fax: SIP Trunking PETER CUTLER SCOTT PAGE November 15, 2011 QUESTIONS AND ANSWERS TODAY S SPEAKERS Peter Cutler Vice President of Sales Instant InfoSystems Scott Page Subject Matter Expert Dialogic
WAN Review A WAN makes data connections across a broad geographic area so that information can be exchanged between distant sites. This topic defines the characteristics of a WAN. WAN Overview WANs connect
Implementing Cisco IOS Telephony and Unified Communications Express (IITUCX) Who should attend The primary audience for this course is as follows: Network administrators Network engineers Systems engineers
POZNAN UNIVE RSITY OF TE CHNOLOGY ACADE MIC JOURNALS No 80 Electrical Engineering 2014 Assaid O. SHAROUN* VOICE OVER IP AND NETWORK CONVERGENCE As the IP network was primarily designed to carry data, it
Traditional Telephony Basic Components of a Telephony Network This topic introduces the components of traditional telephony networks. Basic Components of a Telephony Network 3 A number of components must
Cisco Voice over IP (CVOICE) Q&A DEMO Version Copyright (c) 2010 Chinatag LLC. All rights reserved. Important Note Please Read Carefully For demonstration purpose only, this free version Chinatag study
Overview of Legacy and IP Telephony Networks...3 The Mechanics of Analog and Digital oice Circuits...8 Configuring Router oice Ports...13 CCP COICE Quick Reference Sheets Configuring Router Dial Peers...17
Differences Between FXO and FXS Full Details for Differences Between Foreign Exchange Office (FXO) and Foreign Exchange Subscriber (FXS) Introduction The question What is the difference between FXS and
State of South Carolina Department of Revenue 300A Outlet Pointe Blvd., Columbia, South Carolina 29210 P. O. Box 12265, Columbia, South Carolina 29211 Website Address: http://www.sctax.org SC PRIVATE LETTER
AudioCodes Mediant 800 MSBG Page 1 of 66 6.40A AudioCodes Mediant 800 MSBG 1. Important Notes Check the SIP 3 rd Party Validation Website for current validation status. The SIP 3 rd party Validation Website
Dial Peer Configuration Examples This appendix contains a series of configuration examples featuring the minimum required components and critical Cisco IOS command lines extracted from voice gateway configuration
Case in Point To continue our efforts to help you with your network needs, we will be making several real-world network troubleshooting case studies available to you. The attached case study,, discusses
H.323 / SIP VoIP Gateway VIP GW Quick Installation Guide Overview This quick installation guide describes the objectives; organization and basic installation of the PLANET VIP-281/VIP-480/VIP-880/VIP-1680/VIP-2480
ETM System SIP Trunk Support Technical Discussion Release 6.0 A product brief from SecureLogix Corporation Rev C SIP Trunk Support in the ETM System v6.0 Introduction Today s voice networks are rife with
ACD: Average Call Duration is the average duration of the calls routed bya a VoIP provider. It is a quality parameter given by the VoIP providers. API: An application programming interface (API) is a source
Allstream Converged IP Telephony SIP Trunking Solution An Allstream White Paper 1 Table of contents Introduction 1 Traditional trunking: a quick overview 1 SIP trunking: a quick overview 1 Why SIP trunking?
CHAPTER 2 Sections in this chapter address the following topics: Single Site, page 2-1 Multisite Implementation with Distributed Call Processing, page 2-3 Design Considerations for Section 508 Conformance,
Configuration of Applied VoIP Sip Trunks with the Toshiba CIX40, 100, 200 and 670 Businesses Save Money with Toshiba s New SIP Trunking Feature Unlike gateway based solutions, Toshiba s MIPU/ GIPU8 card
Chapter 2 - The TCP/IP and OSI Networking Models TCP/IP : Transmission Control Protocol/Internet Protocol OSI : Open System Interconnection RFC Request for Comments TCP/IP Architecture Layers Application
Table of Contents About Toshiba Strata CIX and Broadvox SIP Trunking... 1 Requirements... 2 Purpose, Scope and Audience... 3 What is SIP Trunking?... 4 Business Advantages of SIP Trunking... 4 Technical
Curso de Telefonía IP para el MTC Sesión 1 Introducción Mg. Antonio Ocampo Zúñiga Conceptos Generales VoIP Essentials Family of technologies Carries voice calls over an IP network VoIP services convert
Across China: The State Information Center s Nationwide Voice-over-IP Network A customer case study from Quintum Technologies The State Information Center (SIC) is the government body responsible for delivering