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Building Telephony Systems with OpenSIPS 1.6 (e...
20,70 € *
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In DetailSIP is the most important VoIP protocol and OpenSIPS is clearly the open source leader in VoIP platforms based on pure SIP. The whole telecommunication industry is changing to an IP environment, and telephony in the way we know today will disappear in less than ten years. SIP is the protocol leading this disruptive revolution and it is one of the main protocols on next-generation networks. While a VoIP provider is not the only kind of SIP infrastructure created using OpenSIPS, it is certainly one of the most difficult to implement.This book will give you a competitive edge by helping you to create a SIP infrastructure capable of handling tens of thousands of subscribers. You can extend the examples given in this book easily to other applications such as a SIP router, load balancing, IP PBX, and Hosted PBX as well. This book is an update of the title Building Telephony Systems with OpenSER.The book starts with the simplest configuration and evolves chapter by chapter teaching you how to add new features and modules. It will first teach you the basic concepts of SIP and SIP routing. Then, you will start applying the theory by installing OpenSIPS and creating the configuration file. You will learn about features such as authentication, PSTN connectivity, user portals, media server integration, billing, NAT traversal, and monitoring. The book uses a fictional VoIP provider to explain OpenSIPS. The idea is to have a simple but complete running VoIP provider by the end of the book.A practical guide to building an efficient SIP telephony systemApproachThis is a practical, hands-on book based around a fictitious case study VoIP Provider that you will build on a development server using OpenSIPS 1.6. The case study grows chapter by chapter, from installing your local development server, right up to the finished VoIP provider.Who this book is forThis book is for readers who want to understand how to build a SIP provider from scratch using OpenSIPS. It is suitable for VoIP providers, large enterprises, and universities.Telephony and Linux experience will be helpful but is not essential. Readers need not have prior knowledge of OpenSIPS. This book will also help readers who were using OpenSER but are now confused with the new OpenSIPS.

Anbieter: buecher
Stand: 08.08.2020
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Building Telephony Systems with OpenSIPS 1.6 (e...
20,70 € *
ggf. zzgl. Versand

In DetailSIP is the most important VoIP protocol and OpenSIPS is clearly the open source leader in VoIP platforms based on pure SIP. The whole telecommunication industry is changing to an IP environment, and telephony in the way we know today will disappear in less than ten years. SIP is the protocol leading this disruptive revolution and it is one of the main protocols on next-generation networks. While a VoIP provider is not the only kind of SIP infrastructure created using OpenSIPS, it is certainly one of the most difficult to implement.This book will give you a competitive edge by helping you to create a SIP infrastructure capable of handling tens of thousands of subscribers. You can extend the examples given in this book easily to other applications such as a SIP router, load balancing, IP PBX, and Hosted PBX as well. This book is an update of the title Building Telephony Systems with OpenSER.The book starts with the simplest configuration and evolves chapter by chapter teaching you how to add new features and modules. It will first teach you the basic concepts of SIP and SIP routing. Then, you will start applying the theory by installing OpenSIPS and creating the configuration file. You will learn about features such as authentication, PSTN connectivity, user portals, media server integration, billing, NAT traversal, and monitoring. The book uses a fictional VoIP provider to explain OpenSIPS. The idea is to have a simple but complete running VoIP provider by the end of the book.A practical guide to building an efficient SIP telephony systemApproachThis is a practical, hands-on book based around a fictitious case study VoIP Provider that you will build on a development server using OpenSIPS 1.6. The case study grows chapter by chapter, from installing your local development server, right up to the finished VoIP provider.Who this book is forThis book is for readers who want to understand how to build a SIP provider from scratch using OpenSIPS. It is suitable for VoIP providers, large enterprises, and universities.Telephony and Linux experience will be helpful but is not essential. Readers need not have prior knowledge of OpenSIPS. This book will also help readers who were using OpenSER but are now confused with the new OpenSIPS.

Anbieter: buecher
Stand: 08.08.2020
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Plantronics Polycom SoundStation Duo dual-mode ...
988,44 € *
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Das Konferenztelefon SoundStation Duo ist ein hochwertiges Werkzeug zur Produktivitätssteigerung und gleichzeitig ein zuverlässiger Bestandteil Ihrer alltäglichen Arbeitsumgebung. Es eignet sich hervorragend für kleine bis mittelgroße Räume und bietet ein Maximum an Flexibilität, Benutzerfreundlichkeit und Audioqualität.>Highlights:- Ermöglicht Gruppen-Audiokonferenzen in höchster Audioqualität- Im analogen oder IP-Modus verwendbar und mit Online Software-Upgrades spielend leicht aktualisierbar- Kompatibel mit zahlreichen IP-Anrufplattformen für maximale Audioqualität und größtmöglichen Funktionsumfang- Beispiellose Flexibilität: Verbindung zu Mobiltelefonen und PCs zur Interneteinwahl- Einfache Installation und Verwaltung: Dank Online Konfigurationstool kein Boot-Server erforderlich> Switch- PoE 802.3af> Telefoneigenschaften- Telefontyp: Konferenztelefon> IP Telefonie- IP Telefon- SIP-Standard (Session Initiation Protocol)> Systemanschlüsse- Anzahl Ethernet-Ports 10/100 Mbit/s: 1> Audio-Anschlüsse- Headsetanschluss: 2,5 mm Klinke- Lautsprecher: Ja> Display- Displaytechnologie: LED- Beleuchtetes Display- Auflösung Breite: 248 Pixel- Auflösung Höhe: 68 Pixel> Grundeigenschaften- Farbe: Schwarz- Höhe: 34,6 cm- Breite: 32,7 cm- Tiefe: 6,4 cm- Gewicht: 0,74 kg- Produkttyp: Konferenztelefon> Stromversorgung- Strom: 0,5 mA- Eingangsspannung: 100 - 240 V> Technische Beschreibung- Externe Stromversorgung über universelles Netzteil (Wechselstrom): 24 V, 0,5 A, 2,5 mm-Gleichstromstecker- Standardtastatur mit 12 Tasten- Kontextabhängige Softkeys: 4- Auflegen/Abheben, Konferenz, Wahlwiederholung, Stumm, Lauter/Leiser, Menü, Navigationstasten>Audioeigenschaften:- 3 Kardioid-Mikrofone: 200 - 7000 Hz- Lautsprecher-Frequenzgang: 220 - 7000 Hz- 3 m Mikrofonaufnahme- Lautstärke einstellbar bis 86 dB bei 0,5 Metern- Sprechpausenerkennung- Rauschunterdrückung- DTMF-Tonerzeugung/ DTMF-Event-RTP-Payload>Unterstütze Codecs:- G.711 (A-law und µ-law)- G.729a (Annex B)- G.722- iLBC 13,33 und 15,2 kbps>Weitere Schnittstellen:- Analoge RJ-11-PBX- oder FestnetzTelefonschnittstelle mit zwei Kabeln- 2 RJ9-Anschlüsse für Erweiterungsmikrofone>Sicherheit:- Transport Layer Security>Highlights:- Ermöglicht Gruppen-Audiokonferenzen in höchster Audioqualität- Im analogen oder IP-Modus verwendbar und mit Online Software-Upgrades spielend leicht aktualisierbar- Kompatibel mit zahlreichen IP-Anrufplattformen für maximale Audioqualität und größtmöglichen Funktionsumfang- Beispiellose Flexibilität: Verbindung zu Mobiltelefonen und PCs zur Interneteinwahl- Einfache Installation und Verwaltung: Dank Online Konfigurationstool kein Boot-Server erforderlich> Switch- PoE 802.3af> Telefoneigenschaften- Telefontyp: Konferenztelefon> IP Telefonie- IP Telefon- SIP-Standard (Session Initiation Protocol)> Systemanschlüsse- Anzahl Ethernet-Ports 10/100 Mbit/s: 1> Audio-Anschlüsse- Headsetanschluss: 2,5 mm Klinke- Lautsprecher: Ja> Display- Displaytechnologie: LED- Beleuchtetes Display- Auflösung Breite: 248 Pixel- Auflösung Höhe: 68 Pixel> Grundeigenschaften- Farbe: Schwarz- Höhe: 34,6 cm- Breite: 32,7 cm- Tiefe: 6,4 cm- Gewicht: 0,74 kg- Produkttyp: Konferenztelefon> Stromversorgung- Strom: 0,5 mA- Eingangsspannung: 100 - 240 V> Technische Beschreibung- Externe Stromversorgung über universelles Netzteil (Wechselstrom): 24 V, 0,5 A, 2,5 mm-Gleichstromstecker- Standardtastatur mit 12 Tasten- Kontextabhängige Softkeys: 4- Auflegen/Abheben, Konferenz, Wahlwiederholung, Stumm, Lauter/Leiser, Menü, Navigationstasten>Audioeigenschaften:- 3 Kardioid-Mikrofone: 200 - 7000 Hz- Lautsprecher-Frequenzgang: 220 - 7000 Hz- 3 m Mikrofonaufnahme- Lautstärke einstellbar bis 86 dB bei 0,5 Metern- Sprechpausenerkennung- Rauschunterdrückung- DTMF-Tonerzeugung/ DTMF-Event-RTP-Payload>Unterstütze Codecs:- G.711 (A-law und µ-law)- G.729a (Annex B)- G.722- iLBC 13,33 und 15,2 kbps>Weitere Schnittstellen:- Analoge RJ-11-PBX- oder FestnetzTelefonschnittstelle mit zwei Kabeln- 2 RJ9-Anschlüsse für Erweiterungsmikrofone>Sicherheit:- Transport Layer Security(TLS)- Verschlüsselte Konfigurationsdateien- Digest-Authentifizierung- Anmeldung über Kennwort- Unterstützt URL-Syntax mit Kennwort für Boot- Server- Sichere Bereitstellung über HTTPS- Unterstützung von ausführbarer Software mit Signatur- IEEE 802.1x Netzwerkzugangskontrolle> Lieferumfang- Polycom SoundStation Duo dual-mode Netzteil, PIM, PSTN- Telefonkonsole- Kombiniertes Analog- und Ethernet-Kabel mit Stromversorgungsmodul, 6,4 m- Universelles Netzteil: 24 V, 0,5 A- 2,1 m Netzkabel (nach Landesnorm)- 2,1 m Ethernet-Kabel- 2,1 m Telefonkabel (RJ11)- Kurzanleitung

Anbieter: mcbuero
Stand: 08.08.2020
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Asterisk (PBX)
45,00 € *
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High Quality Content by WIKIPEDIA articles! Asterisk is a software implementation of a telephone private branch exchange (PBX) originally created in 1999 by Mark Spencer of Digium. Like any PBX, it allows attached telephones to make calls to one another, and to connect to other telephone services including the public switched telephone network (PSTN) and Voice over Internet Protocol (VoIP) services. Its name comes from the asterisk symbol. Asterisk is released under a dual license model, using the GNU General Public License (GPL) as a free software license and a proprietary software license to permit licensees to distribute proprietary, unpublished system components.

Anbieter: Dodax
Stand: 08.08.2020
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Mobility Management Techniques for Wireless Net...
59,00 € *
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Mobility Management &amp, Route Optimization issues for Wireless Networks are presented in detail. IP-based Networks, PSTN-based Networks and ATM-based Networks backbones have been considered for developing mobility management technique. In IP-based backbone network, a route optimization scheme is proposed to improve the performance of Mobile IP having some apparent drawbacks like triangular routing problem, home agent overloading etc. An encapsulation at one- level up in the hierarchy is performed to improve the handoff delay and the simulation is carried out on NS-2 simulator. In PSTN-based backbone networks, a protocol is proposed and an analytical modeling is carried out where the location registration can be finished prior to the arrival of a MT at the new network to support the inter-system roaming. For an ATM-based backbone network, a design and prototyping of a mobility support ATM network is proposed to improve the handoff time, cell loss and call setup time. Further, an improved mobile location management for wireless ATM networks is proposed and analyzed to improve the performance in terms of number of hops required for location managements.

Anbieter: Dodax
Stand: 08.08.2020
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MAIDS for VoIP
68,00 € *
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Compared to traditional (PSTN) voice networks, a Voice over Internet Protocol network is a convergence of a signaling network and a data network using Internet Protocol (IP). The use of shared media by VoIP systems opens the door to some uncertainty as to the source of a call. While in the traditional voice networks one has to tap into a specific circuit to eavesdrop, in an IP network any equipment connected to the target LAN can identify, store and playback the VoIP packets that traverse that LAN. An unprotected, unauthenticated IP network makes VoIP susceptible to hostile use, such as call hijacking, connection tear down, denial of service, or sending computer viruses over the network. In this work, we perform a series of attacks against a VoIP application, and prove that they succeed with nothing more than a couple of identity tokens captured from the network traffic as prerequisites. We then design an Intrusion Detection System implementing a gradual attack-response procedure, destined to inform and protect the End-Users of the Application Under Test.

Anbieter: Dodax
Stand: 08.08.2020
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IP Phone
29,00 € *
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An IP phone uses Voice over IP technologies allowing telephone calls to be made over an IP network such as the internet instead of the ordinary PSTN system. Calls can traverse the Internet, or a private IP Network such as that of a company. The phones use control protocols such as Session Initiation Protocol, Skinny Client Control Protocol or one of various proprietary protocols such as that used by Skype. IP phones can be simple software-based Softphones or purpose-built hardware devices that appear much like an ordinary telephone or a cordless phone. Ordinary PSTN phones are used as IP phones with analog telephony adapters (ATA).It may have many features an analog phone doesn't support, such as e-mail-like IDs for contacts that may be easier to remember than names or phone numbers.

Anbieter: Dodax
Stand: 08.08.2020
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Voice over internet protocol (VoIP) over WiMAX
49,00 € *
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Existing wireless networks enable different important applications over IP based network such as public internet and Voice over Internet protocol is one of the important applications which have become a possible alternative to public switched telephone network (PSTN). This project investigates one of the time sensitive data applications on WiMAX i.e. VoIP. I carried on WiMAX because WiMAX networks provide advance features and protocols to support the Quality of service (QoS). This project also indicates about the impact of load and mobility on the voice calls. This book provides details about the Voice Over Internet Protocol (VoIP), Which includes types of VoIP calls, VoIP System and Components, VoIP Protocols, VoIP codecs and VoIP Quality of Service (QoS). It also provides detailed study of WiMAX including WiMAX Protocol Layer and WiMAX Quality Of Service(QoS)

Anbieter: Dodax
Stand: 08.08.2020
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Design and Simulation of LAN with GNS3 for a Me...
49,90 € *
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Before now, all telephony services are based on the circuit-switched networks, known as the Public Switched Telephone Networks (PSTN). But nowadays, a new trend has emerged that provides telephony services over internet protocol networks, referred to as Voice over IP (VoIP) or IP telephony. There are a few driving force and motivational factors that are behind the VoIP network - the cost savings, the voice and data applications' integration etc are the crux of the innovation. The cost saving is achieved mainly by reduced cost of making a long distance call, maintenance, deployment, management etc., while the integrated applications of voice and data are teleconferencing, integrated voice mail and email, the call distribution intelligence and automation. Therefor, with this era of innovative business practices in information technology that involves voice and data networks integration, both transmitted over the internet and implemented on a local area network (LAN).This book is undoubtedly beneficial to Small and Medium sized businesses for convergence of data and voice networks in their existing data network to accommodate the VoIP solution in the same LAN.

Anbieter: Dodax
Stand: 08.08.2020
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