Relevant bibliographies by topics / Telephone Signaling

Contents

  1. Journal articles
  2. Dissertations / Theses
  3. Books
  4. Book chapters
  5. Conference papers
  6. Reports

Academic literature on the topic 'Telephone Signaling'

Author: Grafiati
Published: 10 December 2022
Last updated: 28 January 2023

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Journal articles on the topic "Telephone Signaling"

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Mostini, Leonardo. "[Birds trapped in telephone wiring system]." Rivista Italiana di Ornitologia 88, no. 1 (December 5, 2018): 57–58. http://dx.doi.org/10.4081/rio.2018.317.

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[Six cases of birds trapped in telephone wires or in structures connected to telephone wires were signaled in the areas of Turin and Novara (Piedmont, N. W. Italy). The birds stayed hanging fatally either by chance or during a trophic activity. The species involved in the described episodes are five, but become eight when added to the ones subjects of previous signaling: Little bitten Ixobrychus minutus, Grey heron Ardea cinerea, Moorhen Gallinula chloropus, Barn owl Tyto alba, Little owl Athene noctua, Robin Erithacus rubecula, Starling Sturnus vulgaris, Hooded crow Corvus corone cornix.] [Article in Italian]
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Akata, M., Y. Nagataki, K. Koyabu, K. Mukai, S. Yoshida, S. Morisaki, M. Eda, I. Ueki, and T. Matsui. "A no-trimming SLIC two-chip set with coin telephone signaling facilities." IEEE Journal of Solid-State Circuits 25, no. 2 (April 1990): 458–65. http://dx.doi.org/10.1109/4.52170.

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Gluhov, V. T., and M. A. Zharkov. "Toll telephone exchanges and signaling systems in the Russian long-distance network." IEEE Journal on Selected Areas in Communications 12, no. 7 (1994): 1146–51. http://dx.doi.org/10.1109/49.312941.

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Mueller, Milton. "The Switchboard Problem: Scale, Signaling, and Organization in Manual Telephone Switching, 1877-1897." Technology and Culture 30, no. 3 (July 1989): 534. http://dx.doi.org/10.2307/3105950.

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Nishi, H., S. Shinohara, and H. Umemoto. "A robust approach to tele-control using DTMF signaling (telephone call transferring equipment)." IEEE Transactions on Consumer Electronics 37, no. 3 (1991): 572–77. http://dx.doi.org/10.1109/30.85569.

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Lupker, Stephen J., and Gregory J. Fleet. "Callers' Perceptions of Post-Dialing Delays: The Effects of a New Signaling Technology." Proceedings of the Human Factors Society Annual Meeting 31, no. 2 (September 1987): 275–79. http://dx.doi.org/10.1177/154193128703100231.

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One benefit of Common-Channel Signaling (CCS), a new signaling system developed for telephone network applications, would be greatly reduced post-dialing delays. The question addressed was whether unilateral introduction of CCS in either the toll or local network would negatively affect caller behavior in the other network. Results indicate that a CCS toll network would increase caller impatience and abandonments on local calls but not vice versa. Implications of this effect are considered and felt to be far outweighed by the benefits of the CCS system.
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Cui, Jun Xia, and Hu Li Shi. "A Novel Satellite Telephone Communication System Based on SIGSO Satellite." Applied Mechanics and Materials 446-447 (November 2013): 1642–46. http://dx.doi.org/10.4028/www.scientific.net/amm.446-447.1642.

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In recent years, the coverage of terrestrial mobile communication network is more and more extensive, and the mobile phones have occupied the vast majority of voice communications in almost all countries. However, satellite voice communication system is still playing an irreplaceable role on the vast sea and the land area which are not covered by the land mobile communication network. The motion characteristics of the SIGSO satellites and its new demands and characteristics to the system are analyzed, and a novel very low rate satellite telephone system is put forward. The speech coding algorithms are studied, and a suitable speech coding algorithm and coding rate of 600bps is selected for the SIGSO satellite communication system. The overall design of the satellite voice communication network is given, including link budgets, network structure, system components, system signaling and so on. This satellite telephone system can get access to the PSTN network through ground gateway to achieve interoperability with the public network.
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Goldstein, B. S., and L. G. Slutsky. "Distinctive characteristics of call-handling procedures and signaling logic in a Russian public telephone network." IEEE Journal on Selected Areas in Communications 12, no. 7 (1994): 1161–70. http://dx.doi.org/10.1109/49.312944.

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Lechleider, J. W. "Processing Signals by Structured Channel Signaling for High Speed Digital Transmission on Telephone Access Circuits." Digital Signal Processing 3, no. 3 (July 1993): 163–74. http://dx.doi.org/10.1006/dspr.1993.1022.

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Khalifa, Othman O., Raihan Jannati Binti Roslin, and Sharif Shah Newaj Bhuiyan. "Improved voice quality with the combination of transport layer & audio codec for wireless devices." Bulletin of Electrical Engineering and Informatics 8, no. 2 (June 1, 2019): 665–73. http://dx.doi.org/10.11591/eei.v8i2.1490.

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Improving voice quality over wireless communication becomes a demanding feature for social media apps like facebook, whatsapp and other communication channels. Voice-over-internet protocol (VoIP) helps us to make quick telephone calls over the internet. It includes various mechanism which are signaling, controlling and transport layer. Over wireless links, packet loss and high transmission delay damage voice quality. Here VoIP quality will be measured by three main elements which are signaling protocol, audio codec and transport layer. To improve the overall voice quality, we need to combine these three elements properly to get the best score. Otherwise perceptual speech quality will not be the right tool to measure the voice quality. Here we will use Mean Opinion Score (MOS) for calculated jitter values and end to end delay. At the end, best combination of audio codec signaling protocol produced the quality speech.
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Dissertations / Theses on the topic "Telephone Signaling"

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Bajaj, Garima. "A Pre and Post 9-11 Analysis of SS7 Outages in the Public Switched Telephone Network." Ohio University / OhioLINK, 2007. http://rave.ohiolink.edu/etdc/view?acc_num=ohiou1171294573.

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Horwitz, Russel. "An investigation into a DSP implementation of partial response signaling for 4800 bits per second full-duplex data communications over M.1020 telephone lines." Master's thesis, University of Cape Town, 1990. http://hdl.handle.net/11427/8343.

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This thesis investigates high-speed digital transmission over a conditioned, voice-grade telephone circuit (M.1020), using a technique known as partial response signaling, or PRS. In particular, the case where 4800 bps, full-duplex transmission is required in a CCI'PT V. 22 type format is investigated. The main v.22 criterion to be adhered to, is that frequency-division multiplexing (FDM) is to be used as the means of separating thetransmit and receive channels. The carrier frequencies should be 1200 Hz and 2400 Hz respectively. The investigation concerns the modulation and demodulation sections only.
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Cargill, James. "Multitone signalling on telephone lines." Thesis, University of Ottawa (Canada), 1988. http://hdl.handle.net/10393/5543.

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Lu, Tian. "A CORBA-based interface-centric approach to signaling for IP-based telephony services." Thesis, National Library of Canada = Bibliothèque nationale du Canada, 2001. http://www.collectionscanada.ca/obj/s4/f2/dsk3/ftp04/MQ61020.pdf.

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Goulart, Ana Elisa Pereira. "Signaling Architectures for the Interaction of the Session Initiation Protocol and Quality of Service for Internet Multimedia Applications." Diss., Georgia Institute of Technology, 2005. http://hdl.handle.net/1853/6932.

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Interactive multimedia sessions combine requirements of traditional telephony services and Internet applications. This requires call setup, call signaling, negotiation, routing, security, and network resources. Seeking to facilitate the use of quality of service (QoS) mechanisms to users of such applications, this thesis presented new signaling architectures that addressed the interaction of the Session Initiation Protocol (SIP) as the session control signaling protocol and current resource management frameworks. The Differentiated Services (DiffServ) architecture is used as the primary example. The new architectures addressed the roles of SIP agents and proxy servers in subjects such as resource negotiation, call authorization, and end-to-end QoS in heterogeneous networks. First, an architecture based on the use of QoS-enhanced SIP proxies and a SIP-based interface between the application and network layers was developed, implemented in a testbed, and performance enhancements demonstrated. Further studying of the Internet Engineering Task Force (IETF) proposal for the integration of SIP and resource management led to the development of a new signaling scheme, Resource management Overlapped with Answering Delay (ROAD). It explores the SIP user agent interaction with the network in a way that takes advantage of parallel user answering delays and reservation delays. An experimental evaluation of the ROAD scheme showed its call setup delay savings and reduced signaling load. Then, on the interaction of SIP and call admission control, an inter-domain call authorization model that implements the concepts of proxies as gate controllers (QoS-enhanced SIP proxies-GC), and that provides call authorization status and adds more granularity to the authorization process is proposed. This model showed to be scalable in terms of the need to add more resources to compensate for the increasing service load on the servers. Finally, an example framework that applies the new signaling architectures to achieve end-to-end QoS in heterogeneous networks is presented.
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Boutaleb, Tuleen. "Investigation of the effectiveness of digital cellular mobile telephone networks for Trans-European telemetry signalling." Thesis, Glasgow Caledonian University, 2006. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.443185.

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Broström, Anders, and Niclas Kihlstadius. "Prototyp av en VoIP/PSTN-gateway." Thesis, Karlstad University, Division for Information Technology, 2007. http://urn.kb.se/resolve?urn=urn:nbn:se:kau:diva-2646.

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Under de senaste åren har Internettelefonin varit på frammarsch, och i takt med att tekniken mognat har fler och fler börjat se den som ett alternativ till att ringa via telefonnätet. Förutom att det är billigare att ringa över det förstnämnda, så erbjuder Internettelefonin också en rad revolutionerande tjänster. Det är dock troligt att telefonnätet kommer att få tjänstgöra i många år till, och det erbjuder fortfarande överlägset bäst stabilitet och har stor acceptans. Om de två telefoninätverken ska existera sida vid sida, med varsina användarbaser är det lämpligt om de kan fås att samverka, så att användare av det ena kan ringa användare av det andra, och vice versa. Detta kan göras med en VoIP/PSTN-gateway, som översätter kontrollinformation och rösttrafik mellan de två nätverken.

Uppsatsen handlar om det arbete vi har utfört år TietoEnator i Karlstad. Uppgiften bestod i att utveckla en prototyp av en VoIP/PSTN-gateway. Från början var det avsett att systemet skulle klara uppringning från endera en ”vanlig” telefon, eller en så kallad IP-telefon. Därtill skulle rösttrafiken överföras genom ändamålsenlig hårdvara. För att utföra arbetet behövde vi först studera relevanta kommunikationsprotokoll både för telefonnätet och för Internet, för att se hur dessa kunde fås att samverka. Vi behövde också lära oss tillgängliga system, bibliotek och verktyg för att förstå hur vi skulle skapa vårt eget system i den efterkommande implementeringsfasen. På grund av en lång inläsningsperiod och inledande tekniska problem, samt att nödvändig hårdvara för översättning av rösttrafiken inte anlände i tid begränsades arbetet till att innefatta samtal initierade från den vanliga telefonen till ip-telefonen, utan röstöverföring. Likväl har ett resultatgivande arbete utförts, och det beskrivs i detalj i rapporten.


During the past few years Internet telephony has advanced rapidly, and as the technology has evolved, more and more have come to consider it an alternative to making phone calls through the telephone network. Besides being cheaper, Internet telephony also provides several revolutionary services. It is likely though that the telephone network will remain in use for several years to come, and it still offers by far the best stability and is accepted by most people. If the two networks are to coexist, with their respective users, it would be useful if they could be made to interact, so that users of one network can call users of the other, and vice versa. This can be done with a VoIP/PSTN gateway, which translates control information and voice traffic between the two networks.

Our dissertation is about the work we have performed for TietoEnator in Karlstad. The assignment was to develop a prototype of a VoIP/PSTN gateway. Initially the system was meant to support phone calls initiated either from an “ordinary” phone or from an IP telephone. Also the voice traffic was supposed to be translated with the use of appropriate hardware. To manage this we first needed to study all the relevant protocols for communication used in the telephone network and on the Internet, to get an idea of how these could be made to interact. We also had to learn existing systems, libraries and tools in order to see how we could create our own system. Due to a long learning period and technical problems in the beginning, and because the necessary hardware equipment for translation of voice traffic did not arrive in time, the assignment was limited to include only calls initiated from the ordinary phone to the IP telephone, without voice transmission. Never the less, the efforts have produced results, and our work is explained in detail in this dissertation.

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Halamík, Zdeněk. "Možnosti vazby softswitche Asterisk na pobočkové ústředny 4. generace." Master's thesis, Vysoké učení technické v Brně. Fakulta elektrotechniky a komunikačních technologií, 2008. http://www.nusl.cz/ntk/nusl-217329.

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This master’s thesis dissertate the possibilities of the linkage between Asterisk softswitch and the 4th generation private branch exchange. This should create a new generation’s network, so-called NGN, by the convergence of existing telecommunication networks with an IP computer network. This master’s thesis is divided into several chapters. In introduction is described the evolution of the private branch exchanges as well as the principles of the voice digitizing, codecs and signaling commonly used in both TDM and VoIP networks. The main aim of this project is the configuration of Asterisk software exchange for connection with PBX Alcatel 4400 as well as public phone network PSTN. Another goal of this master’s thesis was the configuration of Alcatel PBX and diagnostics of CCS and CAS signaling on E1 interface. In conclusion there are summarized advantages of NGN networks and their utilization in the future.
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Xie, Jiang (Linda). "Mobility Management in Next Generation All-IP Based Wireless Systems." Diss., Georgia Institute of Technology, 2004. http://hdl.handle.net/1853/5190.

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Next generation wireless systems have an IP-based infrastructure with the support of heterogeneous access technologies. One research challenge for next generation all-IP based wireless systems is to design intelligent mobility management techniques that take advantage of IP-based technologies to achieve global roaming between various access networks. To support global roaming, next generation wireless systems require the integration and interoperation of heterogeneous mobility management techniques. Mobility in a hierarchical structure or multilayered environment should be supported. The objective of this study is to develop new mobility management techniques for global roaming support in next generation all-IP based wireless systems. More specifically, new schemes for location management and paging in Mobile IP for network layer mobility support, and new schemes for location management and handoff management in heterogeneous overlay networks for link layer mobility support are proposed and evaluated. For network layer mobility support, a distributed and dynamic regional location management mechanism for Mobile IP is proposed. Under the proposed scheme, the signaling burden is evenly distributed and the regional network boundary is dynamically adjusted according to the up-to-date mobility and traffic load for each terminal. Next, a user independent paging scheme based on last-known location and mobility rate information for Mobile IP is proposed. The proposed scheme takes the aggregated behavior of all mobile users as the basis for paging. For link layer mobility support, an IP-based system architecture for the integration of heterogeneous mobility management techniques is proposed. Three location management schemes under this IP-based architecture are proposed. All the three schemes support user preference call delivery which is a very important feature of next generation wireless communications. A threshold-based enhancement method is also proposed to further improve the system performance. Finally, a hybrid resource allocation scheme for handoff management in wireless overlay networks is proposed. Under this scheme, the overall system resources can be optimally allocated when mobile users are covered by multiple overlay networks.
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Vladimir, Matić. "Novi način procene saobraćajnih svojstava mešovite komunikacione mreže merenjem vremena odziva pozvane strane." Phd thesis, Univerzitet u Novom Sadu, Fakultet tehničkih nauka u Novom Sadu, 2016. http://www.cris.uns.ac.rs/record.jsf?recordId=95785&source=NDLTD&language=en.

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Osnovna svojstva telefonske mreţe, koja opisuju njen kvalitet, su kvalitet prenetog govornog signala i kvalitet u uspostavljanju telefonske veze. Na kvalitet uspostavljanja veze utiĉu mogućnost i brzina uspostavljanja veze. Brzina uspostavljanja veze se izraţava vremenima za koje se obave odreĊene aktivnosti u pojedinim fazama uspo-stavljanja veze. Trajanje ovih faza je propisano normama, koje su odreĊene meĊunarodnim propisima i date su u publikovanim preporukama. Osnovni ĉinilac brzine uspo-stavljanja veze je vreme odziva pozvane strane. Ovaj parametar se definiše kao vremenski interval od slanja poslednjeg adresnog parametra traţenog korisnika od strane pozivajućeg korisnika, do poĉetka odgovora mreţe. U disertaciji je izvršena analiza vremena odziva pozvane strane u telefonskim mreţama i njegovih svojstava. Usled razlika u prenošenju adresnih informacija u klasiĉnim i savremenim paketskim mreţama, uslovi koji se odnose na vreme odgovora pozvane strane nisu isti. Ovaj zakljuĉak je formulisan u predlogu za izmenu postojećih normi koje se odnose na savremene mreţe. U nastavku je opisana mogućnost korišćenja vremena odziva pozvane strane u mešovitoj mreţi kao što je elektroprivredna telefonska mreţa, u cilju donošenja zakljuĉka o radu mreţe. U takvoj mešovitoj mreţi, razliĉite signalizacije i razliĉito trajanje vremena odziva pozvane strane mogu se iskoristiti za nadgledanje ispravnosti pojedinih delova mreţe odnosno detekciju neispravnih veza u njoj. Koristeći ovu osobinu, predloţen je detektor predalarmnog stanja tj. neispravnih veza u mešovitoj telefonskoj mreţi Elektroprivrede Srbije.
Main features of telephone network which describe its service quality are the quality of speech signal and the quality of connection setup. The quality of connection setup is affected by the ability and the speed of connection setup. The speed of connection setup is expressed by the time needed for executing of some connection setup phases. Duration of these phases is prescribed by recommended target values, determined in international recommendations. The key indicator of connection set-up speed is post selection delay. This parameter is defined as the time interval from the moment when the complete address information about the called subscriber is sent from the calling subscriber to the network, till the moment when the answer is received from the network. The analysis of post selection delay in telephone networks and its characteristics is performed in the dissertation. Due to differences in transmission of adress information in classic and modern telephone networks, conditions dealing with the post selection delay are not same. This conclusion is formulated in the proposition for changing existing target values related to modern packet networks. In addition, the possibility of using post selection delay in mixed network, such as the telephone network of electric power utility, for monitoring the network operation is described. In such mixed network, different signaling systems i.e. different durations of post selection delay can be used for checking of proper operation of the network parts and detection of faulty connections. Applying this feature, the pre-alarm state detector is proposed.
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Books on the topic "Telephone Signaling"

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Society, IEEE Communications, ed. Subscriber loop signaling and transmission handbook: Analog. New York: Institute of Electrical and Electronics Engineers, 1992.

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Reeve, Whitham D. Subscriber loop signaling and transmission handbook: Digital. Piscataway, NJ: IEEE Press, 1995.

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Carrel, Harry. Signaling system no. 7: A strategic analysis. New York, NY: Northern Business Information, 1991.

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Downs, Stephen J. Telecommunications networks and signaling system 7. [Washington, D.C.]: Congressional Research Service, Library of Congress, 1992.

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Jeff, Hewett, ed. Signaling System No. 7 (SS7/C7): Protocol, architecture, and services. Indianapolis, IN: Cisco, 2005.

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Lawrence, Harte, Beninger Toni, and Beninger Toni, eds. SS7 basics. 2nd ed. Fuquay-Varina, NC: APDG Pub./Intertec, 2002.

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Phantom traffic: Hearing before the Committee on Commerce, Science, and Transportation, United States Senate, One Hundred Tenth Congress, second session, April 23, 2008. Washington: U.S. G.P.O., 2012.

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Developments in telecommunications: With a focus on SS7 network reliability. Berlin: Springer, 2008.

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Glen, D. V. Networks, signalling, and switching for post-divestiture and the ISDN. [Washington, DC]: Office of the Manager, National Communications System, Technology and Standards Office, 1986.

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Glen, D. V. Networks, signalling, and switching for post-divestiture and the ISDN. [Washington, DC]: Office of the Manager, National Communications System, Technology and Standards Office, 1986.

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Book chapters on the topic "Telephone Signaling"

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Weik, Martin H. "telephone circuit signaling." In Computer Science and Communications Dictionary, 1752. Boston, MA: Springer US, 2000. http://dx.doi.org/10.1007/1-4020-0613-6_19243.

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Zakharova, Larissa. "Electrical Signalling (2): Telephone." In Media and Communication in the Soviet Union (1917–1953), 301–11. Cham: Springer International Publishing, 2022. http://dx.doi.org/10.1007/978-3-030-88367-6_18.

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Pavloski, Mihajlo. "Signalling Attacks in Mobile Telephony." In Communications in Computer and Information Science, 130–41. Cham: Springer International Publishing, 2018. http://dx.doi.org/10.1007/978-3-319-95189-8_12.

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Ackermann, R., V. Darlagiannis, U. Roedig, and R. Steinmetz. "Using DMIF for Abstracting from IP-Telephony Signaling Protocols." In Interactive Distributed Multimedia Systems and Telecommunication Services, 104–15. Berlin, Heidelberg: Springer Berlin Heidelberg, 2000. http://dx.doi.org/10.1007/3-540-40002-8_10.

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"Telephone User Part." In Signaling in Telecommunication Networks, 195–223. New York, USA: John Wiley & Sons, Inc., 2002. http://dx.doi.org/10.1002/0471224154.ch9.

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Noll, A. Michael, and Leo Schenker. "Telephone Signaling Systems, Touch-Tone." In Encyclopedia of Physical Science and Technology, 535–43. Elsevier, 2003. http://dx.doi.org/10.1016/b0-12-227410-5/00766-3.

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"Signaling for Analog Telephone Networks." In Telecommunication System Engineering, 111–37. Hoboken, NJ, USA: John Wiley & Sons, Inc., 2005. http://dx.doi.org/10.1002/0471728489.ch4.

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Welzl, Michael. "New Internet Protocals for Multimedia Transmission." In Multimedia Technologies, 1634–42. IGI Global, 2008. http://dx.doi.org/10.4018/978-1-59904-953-3.ch116.

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This chapter will introduce three new IETF transport layer protocols in support of multimedia data transmission and discuss their usage. First, the stream control transmission protocol (SCTP) will be described; this protocol was originally designed for telephony signaling across the Internet, but it is in fact broadly applicable. Second, UDP-Lite (an even simpler UDP) will be explained; this is an example of a small protocol change that opened a large can of worms. The chapter concludes with an overview of the datagram congestion control protocol (DCCP), a newly devised IETF protocol for the transmission of unreliable (typically real-time multimedia) data streams.
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Funder, John W. "Hormones and Receptors." In Oxford Textbook of Endocrinology and Diabetes 3e, edited by John A. H. Wass, Wiebke Arlt, and Robert K. Semple, 6–12. Oxford University Press, 2021. http://dx.doi.org/10.1093/med/9780198870197.003.0002.

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Hormones are one of the two overarching command and control systems in the body, the other being the central and peripheral nervous system. They complement one another in various ways: hormones are like radio, widely transmitted, requiring a specific receiver (aka receptor); the nervous system is like a landline telephone network. There are obvious differences, in that the nervous system is partially under conscious control. There are less obvious differences, in that endocrine receptors are high affinity, allowing signalling via low levels of circulating hormones; receptors for neurotransmitters are low affinity, rapidly on and off, allowing millisecond responses. After considerations of binding per se, and further distinction between the two control systems, a case study on the complexity surrounding one particular endocrine receptor is presented. The ancillary mechanisms in epithelial tissues conferring aldosterone selectivity on mineralocorticoid receptors, their evolution from a primordial corticoid receptor, and mechanisms of receptor antagonism detailed.
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Fomin, Vladis, and Kalle Lyytinen. "How to Distribute a Cake before Cutting It into Pieces." In Information Technology Standards and Standardization, 222–39. IGI Global, 2000. http://dx.doi.org/10.4018/978-1-878289-70-4.ch014.

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This article analyses social networks by looking at the standard making processes. As a framework for analysis, actor network theory is chosen. Standards are of particular interest for actor network theory for they provide mechanisms to align interests of multiple social groups organized in networks that have a joint incentive in working with the standards and /or associated technologies. These social groups include scientific communities, government institutions and social movements (industrial groups, companies, and consumers) that are interested in regulating and innovating with new technologies. Standards provide the mechanisms to inscribe subsequent behaviors that are expected to become persistent over time. Standard making process is a social process. Actors are involved in the process of continuous negotiation of their interests. Due to this fact, standards became an object of analysis for scholars within the social shaping of technology theory (SST). Though usually scholars of this school take standards as material objects, they interpret technology as such, e.g., a bicycle, or a steam machines. In Information Technology (IT), domain standards are intangible. Those are electronic data exchange formats, communications protocols, signalling protocols, etc. Wireless and mobile communications in particular, being a large field of IT, represent an interesting case for analysis. Present in mobile telephony’s domain are de jure (e.g., GSM) and de facto standards (e.g., NMT). Also the broad scope and large scale of standardization processes suggests non-unified pattern of standard making and complex organizational structure. To make mobile telephony standards successful implies large networks and numerous mandatory passage points. In this paper we apply actor network theory based analysis (ANT) to the development of NMT wireless standards. Researchers interested in IT standardization, except for a few studies on electronic data interchange (EDI) by Hanseth (1997), have overlooked this approach. The acronym NMT stands for Nordisk MobilTelefon (Nordic Mobile Telephone) and it can be historically regarded as one of the best examples of Nordic cooperation in technology as NMT systems have spread quite widely around the world and it also formed an important stepping stone for the evolution of GSM standards. We chose for ANT analysis of the NMT standard making process to learn of the usefulness of theoretical framework and to understand the standard making process of NMT as a social and institutional change. In our opinion, this more than anything else, explains the success of this interesting historical incident that changed the telecommunication industry radically and made Scandinavia a powerhouse of the wireless technologies. Our approach expected to bring more understanding on how the enthusiasm of a small number of actors fostered successful development of the NMT cellular telephony standard. At the same time the NMT standard was based on concepts and visions of its developers. Yet, it was these visions and engagements that lead to distributed the big cake of the cellular world even before cutting it into pieces. The outline of the chapter is the following. In the next section, we discuss past theoretical analysis of the topic. Then we introduce new notions into ANT, such as a layer and a multilayered structure. Next we tell the story of the Nordic radio engineers’ gang. We then analyze the NMT standard’s development process as an instance of actor network mobilization. Some insights into future developments of cellular mobile communications, both from the technological and social perspectives are provided.
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Conference papers on the topic "Telephone Signaling"

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Akata, M., Y. Nagataki, K. Koyabu, K. Mukai, S. Yoshida, and I. Ueki. "A no-trimming SLIC two-chip set with coin telephone signaling facilities." In 1989 Proceedings of the IEEE Custom Integrated Circuits Conference. IEEE, 1989. http://dx.doi.org/10.1109/cicc.1989.56771.

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Pavloski, Mihajlo, and Erol Gelenbe. "Signaling Attacks in Mobile Telephony." In International Conference on Security and Cryptography. SCITEPRESS - Science and and Technology Publications, 2014. http://dx.doi.org/10.5220/0005019802060212.

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Dalgic, Ismail, and Hanlin Fang. "Comparison of H.323 and SIP for IP telephony signaling." In Photonics East '99, edited by Andrew G. Tescher, Bhaskaran Vasudev, V. Michael Bove, Jr., and Barbara Derryberry. SPIE, 1999. http://dx.doi.org/10.1117/12.371194.

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Geneiatakis, Dimitris, and Costas Lambrinoudakis. "A Cost-Effective Mechanism for Protecting SIP Based Internet Telephony Services Against Signaling Attacks." In 4th International ICST Mobile Multimedia Communications Conference. ICST, 2008. http://dx.doi.org/10.4108/icst.mobimedia2008.3857.

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Kouroshnezhad, Sahar, Saadan Zokaei, Hassan Yeganeh, and Mohammad SayadHaghighi. "Slot Reservation Demand Assignment Multiple Access Control Protocol For Signalling Of Telephony Traffic Via Geo Satellite." In 2008 10th International Conference on Advanced Communication Technology. IEEE, 2008. http://dx.doi.org/10.1109/icact.2008.4493855.

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Reports on the topic "Telephone Signaling"

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Livingood, J., and R. Shockey. IANA Registration for an Enumservice Containing Public Switched Telephone Network (PSTN) Signaling Information. RFC Editor, November 2006. http://dx.doi.org/10.17487/rfc4769.

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Schulzrinne, H., and T. Taylor. Definition of Events for Channel-Oriented Telephony Signalling. RFC Editor, June 2008. http://dx.doi.org/10.17487/rfc5244.

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Coene, L., and J. Pastor-Balbas. Telephony Signalling Transport over Stream Control Transmission Protocol (SCTP) Applicability Statement. RFC Editor, February 2006. http://dx.doi.org/10.17487/rfc4166.

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