Jaina Systems United States

Jaina Systems a nation wide wholesale Telecom carrier company was established in May 2002, with its main focus to deliver Quality Voice Termination world wide to carriers and calling card providers. Since its inception Jaina System has build up its carrier connectivity network through out Europe and North America and today it is connected to major number of European and American Carrier which gives an opportunity for our wholesale clients to connect any major Carrier through the Jaina Network. With our presence in all major European carrier hotels via SDH network, our clients enjoy their choice of providers, quality and price.With Jaina Systems "Inbound IP & Outbound TDM" policy wholesalers and retailers virtually get up and running the same day without compromising their quality. Jaina maintains a very high level of voice quality on the "Inbound IP" by keeping a very low latency, packet loss, post dial delay. Unlike conventional IP networks the Quality of Jaina`s Network is monitored round the clock by its own and operated INOC at Long Island. Jaina has also introduced first of its kind voice quality measurement standards based on PAMS and PSQM metrics that is acknowledged by the Industry for voice quality measurement.

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General office

Jaina Systems
235 Hilliside Avenue
Williston Park
New York ,11596
United States

Phone: +1-516-741-3027
Site: http://www.jainasystems.com

Contact Person

Surajit Bose / System Engineer
Phone: +1-516-741-3027

International VoIP Wholesale Provider

Call Routing VoIP Solutions
Gatekeepering
H.323 Wireless/ GSM VoIP Solutions
Partnering for Origination and Termination
Phone To Phone
Project Mangement Services
System Integration
Termination

Cisco Systems
Quintum

VoiceMaster

GalaxieTalk.com

InZon Corp

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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