Al Hussaini Telecom Jordan

Al- Hussaini Telecom is a global carrier and provider of high quality international long distance telecommunication services, exclusively sells to other telecommunication companies and has strategically decided two target high yield niche markets in Asia and Africa.

We offer prepaid calling services, residential and corporate accounts and international traveler''s calling card for selective destinations.

We are also Internet Service Provider, licensed from The Telecommunication Regulatory Commission.

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General office

Al Hussaini Telecom
Amman -Jordan
AMMAN
,541010
Jordan

Phone: 00962 6 5605611
Phone: 00962 6 5605644
Site: http://www.al-hussainitelecom.com

Contact Person

Yousef Al-Hussaini / Chief Executive Officer
Phone: 00962 6 5605611
Phone: 00962 79 5571320

Hardware
Hosted VoIP billing service provider
Incumbent PTT
International VoIP Wholesale Provider
Internet Telephony Service Provider
Internet VoIP and Video Conferencing Service Provider
Network Service Provider
SIP Billing
VoIP Billing Software Provider
Voip consulting
Voip Termination ISP
Wireless Broadband

Billing Software
Call Relay VoIP Solutions
Call Routing VoIP Solutions
Collaboration
Consultancy Services
Data Conferencing
E1 Bulk Wholeseller
Fax To Fax
Gatekeepering
H.323 Softswitch & CPE
H.323 VoIP Gateway
H.323 Wireless/ GSM VoIP Solutions
Installation and Support Services

Cisco Systems
Huawei VoIP Products
Lucent
net.com
Netvox
Nicstel
Nuera
Quintum

AA VoIP Enabler
Billing system of own development
Custom Open Source Billing Solution
Dialogue Billing
ETS Billing
Java Based own system
KSN billing
NACT - UNIX Platform
NTS Billing
Oracle Based own system
PortaSIP
Telco Solutions Billing System
TeleCount Wholesale Billing
VoiceMaster
VPAN

Network Exchange Technology

Arab American Telecom LLC (AAT)

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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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