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Gccom France

GCCOM offers affordable and reliable VoIP solutions for the small and large business alike, using same quality and reliability already experienced for voice and fax through local telephone companies.

GCCOM offers affordable and reliable VoIP solutions for the small and large business alike, using same quality and reliability already experienced for voice and fax through local telephone companies.

Our Mission

Is to deliver the most advanced, quality-driven telecommunications and Internet solutions to our customers, including voice over IP (VoIP) products and services such as calling cards and broadband telephony.

VoIP Solutions

* Scalable and flexible for any size business
* Cost Effective inneroffice and remote location calling
* Fixed low monthly rates
* 75% saving on long distance calls
* Connect remote sites and home users to an existing network
* Reliable service with high voice quality

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General office

Gccom
Paris
Paris
France ,75008
France

Phone: 33145759833
Site: http://www.gccom.fr
  • International VoIP Wholesale Provider
  • Network Service Provider
  • Voip consulting
  • Voip Termination ISP
  • Call Relay VoIP Solutions
  • Call Routing VoIP Solutions
  • Collaboration
  • Consultancy Services
  • E1 Bulk Wholeseller
  • H.323 Softswitch & CPE
  • H.323 VoIP Gateway
  • Installation and Support Services
  • IP devices
  • PC to Phone
  • Phone To PC
  • Phone To Phone
  • Project Mangement Services
  • SIP Softswitch & CPE
  • SIP VoIP Gateway
  • System Integration
  • Termination
  • Voice and Video Conferencing
  • Web Call
  • Web To Phone
  • Cisco Systems
  • net.com
  • Billing system of own development
Ing Telecom
Telemedia Communications
Providers in database: 6433
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VoIP Favors SIP

VoIP Favors SIPSession Initiation Protocol (SIP) is a text-based standard put in place by the Internet Engineering Task Force (IETF) and is the primary protocol for Voice over Internet Protocol services. SIP is similar to a MGCP in that it's a signaling protocol that can create, maintain and terminate sessions in IP-based networks. These sessions include multimedia conferencing and two-way phone calls. SIP protocol can run on User Datagram Protocol (UDP), Stream Control Transmission Protocol (SCTP) and Transmission Control Protocol (TCP) because it was created to function independently from the underlying Transport Layer as an Application Layer.

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