LEAD TELECOM CORP. United States

LEAD TELECOM is a wholesale VoIP carrier incorporated in Miami (Florida), with headquarters in the state of Florida, and collocation facilities in New York – EQUINIX and Miami - NAP

LEAD TELECOM started providing voice services in 2010. Innovative strategies and reduced processes management have helped to be recognized as a vital partner by major Tier 1 carriers and mobile operators around the world it was interconnected with. Today Lead Telecom holds wholesale licenses in USA and offers direct termination globally through its worldwide network of direct interconnections and carries.

LEAD TELECOM currently offers a complete suite of solutions ranging from wholesale termination, to our reseller platform offering white label solution for Voip termination, Voip Origination, DID''s and SIP Trunking, among other services. Our philosophy is based on quality. We understand that our clients (carriers, distributors,resellers, calling card operators, and white label users) make more money if the end users talk longer.

LEAD TELECOM is dedicated to make each customer as successful as it can be. Our voice services offer unparalleled reliability, security, and value and our outstanding customer service will leave you wondering why you ever trusted your business to anyone else. We welcome you to explore a business relationship between Lead Telecom and your company regarding Voice Traffic Termination and other Carrier Services, Enterprise Solutions, and more...

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General office

LEAD TELECOM CORP.
85 Grand Canal Dr., Suite # 406
Miami
FL ,33144
United States

Phone: +1 305 468 4643
Site: http://www.leadtelecom.com

International VoIP Wholesale Provider
Wholesale VoIP Carrier

Partnering for Origination and Termination
Sip Trunking
Termination

Cisco Systems
Nextone MSC 4

Billing system of own development

FirstOne Telecom LLC

VoIP Corporation

Providers in database: 6433

RTP Delivers Multimedia Data In Real Time

A 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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