C.A.TELECOM Sistemas de Telecomunicações Lda Portugal

WebPhone, a provider of voice over Internet Protocol (VoIP) technology, was founded in 2002. WebPhone provides PC2Phone calling service using a worldwide VoIP network. The company is based in Portugal and recently opened sales and customer service offices WebPhone, a provider of voice over Internet Protocol (VoIP) technology, was founded in 2007. WebPhone provides PC2Phone calling service using a worldwide VoIP network. The company is based in Portugal and recently opened sales and customer service offices in the United States of America. WebPhone routes a call from your computer over the Internet and terminates the call on a standard telephone using state of the art VoIP technology. The result is high-quality VoIP calls to any phone in the world. By routing the calls over the Internet, WebPhone incurs lower costs than a standard telephone company and is able to pass the cost savings along to you, . The WebPhone service features an online account center for all of your account management needs and an easy-to-use software program, which is free to download.

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General office

C.A.TELECOM Sistemas de Telecomunicações Lda
Rua São Francisco nº 6 A
Elvas
Alentejo ,7350-425
Portugal

Phone: +351 268623225
Site: http://www.catelecom.pt
Contact Person

Carlos Gago / Eng.
Phone: +351 268623225 Opção
  • Hosted PBX Service Provider
  • Hosted VoIP billing service provider
  • International VoIP Wholesale Provider
  • Internet Fax Service Provider
  • Internet Telephony Service Provider
  • Mobile VoIP
  • Network Service Provider
  • SIP Billing
  • VoIP Billing Software Provider
  • Voip consulting
  • Voip Termination ISP
  • Wholesale VoIP Carrier
  • Wireless Broadband
  • Call shop solution
  • Cisco Systems
  • GSM-Central

Providers in database: 6433
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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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