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H.323
35,00 € *
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High Quality Content by WIKIPEDIA articles! H.323 is an umbrella Recommendation from the ITU Telecommunication Standardization Sector (ITU-T) that defines the protocols to provide audio-visual communication sessions on any packet network. The H.323 standard addresses call signaling and control, multimedia transport and control, and bandwidth control for point-to-point and multi-point conferences. It is widely implemented by voice and videoconferencing equipment manufacturers, is used within various Internet real-time applications such as GnuGK and NetMeeting and is widely deployed worldwide by service providers and enterprises for both voice and video services over Internet Protocol (IP) networks. It is a part of the ITU-T H.32x series of protocols, which also address multimedia communications over Integrated Services Digital Network (ISDN), Public Switched Telephone Network (PSTN) or Signaling System 7 (SS7), and 3G mobile networks. H.323 Call Signaling is based on the ITU-T Recommendation Q.931 protocol and is suited for transmitting calls across networks using a mixture of IP, PSTN, ISDN, and QSIG over ISDN.

Anbieter: Dodax AT
Stand: 26.01.2020
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H.323
34,00 € *
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High Quality Content by WIKIPEDIA articles! H.323 is an umbrella Recommendation from the ITU Telecommunication Standardization Sector (ITU-T) that defines the protocols to provide audio-visual communication sessions on any packet network. The H.323 standard addresses call signaling and control, multimedia transport and control, and bandwidth control for point-to-point and multi-point conferences. It is widely implemented by voice and videoconferencing equipment manufacturers, is used within various Internet real-time applications such as GnuGK and NetMeeting and is widely deployed worldwide by service providers and enterprises for both voice and video services over Internet Protocol (IP) networks. It is a part of the ITU-T H.32x series of protocols, which also address multimedia communications over Integrated Services Digital Network (ISDN), Public Switched Telephone Network (PSTN) or Signaling System 7 (SS7), and 3G mobile networks. H.323 Call Signaling is based on the ITU-T Recommendation Q.931 protocol and is suited for transmitting calls across networks using a mixture of IP, PSTN, ISDN, and QSIG over ISDN.

Anbieter: Dodax
Stand: 26.01.2020
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Plain Old Telephone Service
40,10 € *
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Please note that the content of this book primarily consists of articles available from Wikipedia or other free sources online. Plain old telephone service (POTS) is the voice-grade telephone service that remains the basic form of residential and small business service connection to the telephone network in many parts of the world.The name is a retronym, and is a reflection of the telephone service still available after the advent of more advanced forms of telephony such as ISDN, mobile phones and VoIP. POTS has been available almost since the introduction of the public telephone system in the late 19th century, in a form mostly unchanged to the normal user despite the introduction of Touch-Tone dialing, electronic telephone exchanges and fiber-optic communication into the public switched telephone network (PSTN).

Anbieter: Dodax AT
Stand: 26.01.2020
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YATE Yet Another Telephony Engine
50,40 € *
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Please note that the content of this book primarily consists of articles available from Wikipedia or other free sources online. Yate is a telephony engine, a softswitch with PBX capabilities, originally created in 2004 by Paul Chitescu of NullTeam. It is released under a GNU General Public License with an exception for linking with OpenH323 and PWlib (licensed under MPL). Started as a free/open source softswitch for connecting PSTN with VoIP, support for other telephony technologies like conference, PBX, IVR has been added later on. During the years Yate has evolved into a Unified Communications server providing advanced integration with instant messaging, video and fax. Historically the basic function of Yate is the softswitch, connecting together call legs either to external devices or internal server resources. The logic for routing calls uses messages and can be implemented in configuration files, database queries or external programs. Call signalling is possible using PSTN protocols like ISDN or SS7, either directly attached over a TDM interface card (T1, E1, BRI), analog interface or by a remote Signaling gateway. A variety of VoIP protocols can be used without the need of special hardware: SIP, H.323, IAX2 or Jingle

Anbieter: Dodax AT
Stand: 26.01.2020
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YATE Yet Another Telephony Engine
49,00 € *
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Please note that the content of this book primarily consists of articles available from Wikipedia or other free sources online. Yate is a telephony engine, a softswitch with PBX capabilities, originally created in 2004 by Paul Chitescu of NullTeam. It is released under a GNU General Public License with an exception for linking with OpenH323 and PWlib (licensed under MPL). Started as a free/open source softswitch for connecting PSTN with VoIP, support for other telephony technologies like conference, PBX, IVR has been added later on. During the years Yate has evolved into a Unified Communications server providing advanced integration with instant messaging, video and fax. Historically the basic function of Yate is the softswitch, connecting together call legs either to external devices or internal server resources. The logic for routing calls uses messages and can be implemented in configuration files, database queries or external programs. Call signalling is possible using PSTN protocols like ISDN or SS7, either directly attached over a TDM interface card (T1, E1, BRI), analog interface or by a remote Signaling gateway. A variety of VoIP protocols can be used without the need of special hardware: SIP, H.323, IAX2 or Jingle

Anbieter: Dodax
Stand: 26.01.2020
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Plain Old Telephone Service
39,00 € *
ggf. zzgl. Versand

Please note that the content of this book primarily consists of articles available from Wikipedia or other free sources online. Plain old telephone service (POTS) is the voice-grade telephone service that remains the basic form of residential and small business service connection to the telephone network in many parts of the world.The name is a retronym, and is a reflection of the telephone service still available after the advent of more advanced forms of telephony such as ISDN, mobile phones and VoIP. POTS has been available almost since the introduction of the public telephone system in the late 19th century, in a form mostly unchanged to the normal user despite the introduction of Touch-Tone dialing, electronic telephone exchanges and fiber-optic communication into the public switched telephone network (PSTN).

Anbieter: Dodax
Stand: 26.01.2020
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Patton SmartNode 4324
1.125,06 € *
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- 16, 24 or 32 FXS or FXO—Simultaneous voice or fax calls on all ports. Advanced local call switching.- Full SIP and T.38 support—Supports the complete range of industry standard VoIP: SIP, H.323, T.38 fax, fax and modem bypass, DTMF relay. Codecs G.729, G.723 etc.- Outstanding Interoperability—Interoperable for voice and T.38 fax with leading SIP service providers, soft-switch vendors and Asterisk IP-PBXThe SmartNode 4300 VoIP Gateway provides 12 to 32 analog FXS or FXO interfaces to connect phones, fax or PSTN trunk lines to your IP-based communications (IP PBX, UC systems and SIP Trunks).Like every SmartNode, the SN4300 supports every industry-standard CODEC to deliver toll-quality voice on every call. The Unified Communications Agent™ (UCA) provides any-to-any multi-path switching (simultaneous SIP, H.323, ISDN, and POTS calls with routing and conversion between TDM/PSTN and IP/Ethernet networks—plus T.38 and SuperG3 FAX). VoIP-over-VPN with voice encryption provides secure voice and data via IPsec with AES/DES strong encryption and automated keying via Internet Key Exchange (IKE). In addition, advanced call-router functionality includes least-cost call routing with flexible dialed-number plan support.The SmartNode survivability suite provides PSTN fallback to ensure business continuity in case the IP network fails. In addition, SmartNode delivers a smooth transition to VoIP with strong number portability support accepting incoming calls from the PSTN throughout the VoIP service provider's number porting process.Preserve investments in legacy phone equipment while taking the next steps toward unified communications with the SN4300 VoIP Gateway. Providing 12 to 32 FXS or FXO interfaces and one 10/100/1000 Ethernet, the SN4300 delivers a reliable, cost-effective solution for the Enterprise.

Anbieter: Dodax AT
Stand: 26.01.2020
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Patton SmartNode 4324
1.125,89 € *
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- 16, 24 or 32 FXS or FXO—Simultaneous voice or fax calls on all ports. Advanced local call switching.- Full SIP and T.38 support—Supports the complete range of industry standard VoIP: SIP, H.323, T.38 fax, fax and modem bypass, DTMF relay. Codecs G.729, G.723 etc.- Outstanding Interoperability—Interoperable for voice and T.38 fax with leading SIP service providers, soft-switch vendors and Asterisk IP-PBXThe SmartNode 4300 VoIP Gateway provides 12 to 32 analog FXS or FXO interfaces to connect phones, fax or PSTN trunk lines to your IP-based communications (IP PBX, UC systems and SIP Trunks).Like every SmartNode, the SN4300 supports every industry-standard CODEC to deliver toll-quality voice on every call. The Unified Communications Agent™ (UCA) provides any-to-any multi-path switching (simultaneous SIP, H.323, ISDN, and POTS calls with routing and conversion between TDM/PSTN and IP/Ethernet networks—plus T.38 and SuperG3 FAX). VoIP-over-VPN with voice encryption provides secure voice and data via IPsec with AES/DES strong encryption and automated keying via Internet Key Exchange (IKE). In addition, advanced call-router functionality includes least-cost call routing with flexible dialed-number plan support.The SmartNode survivability suite provides PSTN fallback to ensure business continuity in case the IP network fails. In addition, SmartNode delivers a smooth transition to VoIP with strong number portability support accepting incoming calls from the PSTN throughout the VoIP service provider's number porting process.Preserve investments in legacy phone equipment while taking the next steps toward unified communications with the SN4300 VoIP Gateway. Providing 12 to 32 FXS or FXO interfaces and one 10/100/1000 Ethernet, the SN4300 delivers a reliable, cost-effective solution for the Enterprise.

Anbieter: Dodax
Stand: 26.01.2020
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beroNet Modular Session Border Controller BNSBC...
1.498,21 € *
zzgl. 4,70 € Versand

- Gerätetyp: VoIP-Gateway - Formfaktor: Rack-montierbar - 1U - Data Link Protocol: Ethernet, Fast Ethernet, TDM - Kapazität: Max. gleichzeitige PSTN-Kanäle: 64 ¦ Max. gleichzeitige Sessions: 32 - Telefonieschnittstellen: 1 Leitung (ISDN T-1/E-1) - Digital Signaling Protocol: SIP - Protokolle und Spezifikationen: ITU V.29, ITU V.17, ITU V.27ter, ITU V.27bis, ITU T.38 - Anzahl der digitalen Anschlüsse: 1 - Stromversorgung: Wechselstrom 120/230 V - Abmessungen (Breite x Tiefe x Höhe): 34 cm x 25.5 cm x 4.6...

Anbieter: Schäfer Shop
Stand: 26.01.2020
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