Kernel Network Transit Telecom s.r.o. Czech Republic

Kernel Network was founded in 2001. Today Kernel Network is a universal operator providing SIP-telephony services in the corporate and consumer markets. The group of companies Kernel Network provides since 2002 the services of international telecommunicat

Kernel Network was founded in 2001. Today Kernel Network is a universal operator providing SIP-telephony services in the corporate and consumer markets. The group of companies Kernel Network provides since 2002 the services of international telecommunication, using it`s own technologies of the voice and facsimile information transfer.

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General office

Kernel Network Transit Telecom s.r.o.
2191 Janskeho
Prague
,15500
Czech Republic

Phone: + 420 224 500 701
Phone: + 420 608 184 962
Site: http://www.freeopencall.cz
Contact Person

Anton Skorokhod / VoIP ing.
Phone: 00 420 224 500 702
Phone: 00 420 608 551 734
  • Hardware
  • Hosted VoIP billing service provider
  • International VoIP Wholesale Provider
  • Internet Telephony Service Provider
  • SIP Billing
  • VoIP Billing Software Provider
  • Voip Termination ISP
  • Billing Software
  • Call Relay VoIP Solutions
  • Call Routing VoIP Solutions
  • E1 Bulk Wholeseller
  • Fax To Fax
  • Gatekeepering
  • H.323 Softswitch & CPE
  • H.323 VoIP Gateway
  • Installation and Support Services
  • IP devices
  • Outsourced Billing
  • Partnering for Origination and Termination
  • PC to Phone
  • Phone To PC
  • Phone To Phone
  • SIP Softswitch & CPE
  • SIP VoIP Gateway
  • System Integration
  • Termination
  • Web To Phone
  • Cisco Systems
  • Dialogic Based
  • another
  • Billing system of own development

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RTP Delivers Multimedia Data In Real Time

RTP Delivers Multimedia Data In Real TimeA Real-time Transport Protocol (RTP) is a protocol standard defined by the Internet Engineering Task Force (IETF) that outlines a management system for programs using real-time transmission of data through mutlicast or unicast network services in multimedia situations. Originally designed by the IETF for videoconferencing for multiple participants, the protocol is largely used in Voice over Internet Protocol applications. Despite its name, the protocol cannot guarantee real-time delivery of data, however, the RTP does compensate for jitters and can detect when data arrives out of sequence, both issues being popular in VoIP communication. In IP telephony, RTP works with a signaling protocol, such as SIP or H.323, in order to set up connections in a network.

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