- This topic has 4 replies, 1 voice, and was last updated 22 years, 3 months ago by pram.
13th June 2001 at 06:28 #20435Asif SiddiqueGuest
I have an assignment to under stand what is voice/ip Please send me relavent material
Thanks20th June 2001 at 04:07 #20436TomGuest
Check out Cisco’s web site.21st June 2001 at 10:03 #20437voipGuest
we’ve been doing the same for last 2 months and have made a document on Voip for beginners.If u r interested.
email@example.com June 2001 at 00:47 #20438AdifGuest
I enclose some information which might be helpfull to you. This 4 port gatwaye FXO/FXS will cost you $1300.
Let me know if you have a server or you will be using a third party server.
IP1000 is a high performance industry standard-compliant gateway optimized for VoIP and IP-PBX applications. It is the latest addition to Internet Telephony products. Inheriting proud tradition of technlogical innovation, IP1000 is desgined to possess many sensible features.
The modularized FXO/FXS dual telephone interface provides the system with ultra flexible PSTN connection.IP1000 can conveniently connect to either the trunk or the extension of a PBX, or a mix of both. Direct connections from C.O. lines or analog telephone sets can also be done by IP1000.
In addition, the state-of-the-art design provides users of IP1000 two usage modes – VPN mode and ITSP mode.
In VPN mode, IP1000 constructs a point-to-point communication structure in a Private Network or a Virtual Private Network and functions perfectly in any TCP/IP environment. Also, with a specially designed user-definable dial plan function, IP1000 delivers an optimized user-friendly interface while satisfying the unique dialing habits of each IP1000 user. This enables decentralized enterprises to make phone calls to branches easily and efficiently. With an IP1000 installed in the office, even oversea calls will sound and cost similar to those between internal extensions.
In ITSP mode, IP1000 works as a multi-ports Internet-phone or box, allowing it to reach or be reached directly by all DSG VoIP devices, dial-up and Ethernet-based. Users pf IP1000 can be enjoy the benefits of a gatekeeper and billing system, supported through global VoIP network. Moreover, connecting to the legacy PSTN with just a single IP1000 is also supported. IP1000 is equipped with a RJ-45 female connector interface to the IEEE 802.3 10BaseT Ethernet. With up to 8 analog ports and supports both FXO and FXS telephone interface, IP1000 is more than a reliable and affordable long distance saving phone system, it is the ideal mate for decentralized or small to medium size enterprises.
IP1000 is built to incorporate many user-friendly features. The Telnet setup guide provides a clear tour to configure IP1000. By employing the user-definable dial plan option, making calls through IP1000 is easy and efficient.
IP1000 possesses its own processor and network interface; it works intelligently. No computer, no software and no long-distance carrier required with IP1000.
Clear Voice Quality
IP1000 is designed with the latest and state-of-the-art compression technology, fully compliant with industry standards. Incorporated with advanced echo cancellation algorithm and ultra-low latency jitter control, IP1000 delivers the best voice quality to users.
As a multi-port VoIP gateway, IP1000 functions as a bridge between the call initiator and receiver. This enables users of IP1000 to connect under the following modes: IP1000-to-IP1000, IP1000-to-Phone, Phone-to-IP1000, Phone-to-Phone, IP1000-to-Device and Device-to-IP1000.
Each VoIP products owns a unique ID number. Simply dial the ID of receiving device and one will be connected to other on-line VoIP product users automatically. This enables IP1000 to perform under these modes: IP1000-to-Phone, Phone-to-IP1000, IP1000-to-Device, Device-to-IP1000.
Dimensions 315mm X 198mm X 60mm
Weight Approximately 2300g
Power Input DC 5V,12V
Temperature 0 – 50° C ( 32° ~ 122° F )
Humidity 10% to 90% non-condensing
Internal RAM 2MB / 4MB
Flash Memory 4Mbits standard, 8Mbits optional
Voice Compression G.723.1 standard, G.711, G729a Optional
Telephone Interface FXO / FXS, RJ11
Network Interface Ethernet 10BaseT, RJ45
Network Protocol TCP/IP, DHCP, Telnet
Further information on the server if you like to run your own ITSP Server which cost $22000.
IP2000 is an IP Telephony gateway making phone calls through the Internet. It is the leading edge communication system that provides great opportunities to improve competition and generate huge savings.
It allows a phone device to place calls to any other phone devices covered within the service areas. IP2000, an important role of InterPhoneWork, is a high performance Internet telephony gateway system that provides voice connectivity to other Internet telephony equipments in InterPhoneWork via TCP/IP networks.
IP2000 Virtual Private Network (VPN) Release
It’s an excellent solution for enterprises to reduce or eliminate the long distance phone charges. It can provide phone-to-phone communication service through IP network by bridging the traditional circuit-switched telephony world and Internet/Intranet. Being able to integrate with any telephone system virtually, IP2000 functions effectively in corporate environments. Business users can therefore access IP2000 from any touch-tone phones to place long distance calls over the Internet at the cost of Internet connection only.
IP2000 Internet Telephony Service Provider (ITSP) Release
IP2000 also provides a wide range of features for the ITSP, known as IP2000 ITSP Option Pack. This package includes advanced billing support, easy-to-configure call-routing system, and channel management functions. The ITSP administrator can easily monitor the status and traffic of each channel in real-time. With all superior managerial functions and high quality voice transmissions, IP2000 is the best solution for ITSPs who are interested in Internet telephony services.
IP2000 ITSP Option Pack:
The Billing System includes:
— a main billing server
— a report generator
— a database migrator, and
— an on-line ITSP-account authorizing/managing site on the Web.
The Routing System:
— is able to setup different rates for various traffics (see InterPhoneWork)
— enables the calling time slice be minimized up to 1 second
— optimizes the resolution of the defined rates to only 0.01 cent(US$)
— has an intuitive and easy-to-configure interface
Dynamic Call Routing
Each call is routed to the appropriate remote IP2000 depending on the destination analysis, and offered the least expensive way of completing the call; or if that IP2000 is unavailable, the call is routed to other auxiliary IP2000s.
Advanced Echo Cancellation
A sophisticated echo cancelling algorithm which can supply both line and acoustic echo cancellation.
Real-Time Full Duplex
IP2000 supports full duplex operation; both parties can speak at the same time with clear sound quality.
Dynamic Jitter Buffer
Cooperated with the voice compression technology, IP2000 collects a sufficient number of inbound packets so that playback can be continuous, even though the time between packet arrival is highly variable. In the mean time, IP2000 models cross-network performs and adapts to the size of the jitter buffer accordingly.
With the dedicated board, IP2000 minimizes both the usage of CPU power and voice data processing time. Also IP2000 use a state-of-the-art NT real-time subsystem so that the latency maintains at an optimized condition.
IP2000’s Peak Threshold Adjustment offers users to install the lowest speech volume. If the speech volume falls below the threshold, IP2000 will detect the speech as silence. The gateway will not transfer the voice data during the period of silence; therefore, the bandwidth can be saved significantly.
The well-designed interface blocks the unauthorized incoming and outgoing calls, restricts invalid user ID to access the system, and IP2000 secures the system from any abusive performance.
High Quality of Voice
IP2000 technology improves packet loss recovery, reduces latency and delivers the best voice quality within the lowest bandwidth. The MOS (Mean Opinion Score) can reach as high as 3.98 in 6.3kbps compression rate.
Flexible Billing System
IP2000’s billing system supports both centralized and distributed billing modes. The system tracks every line and records the status of each call such as date, time, source, destination, and duration.
The underlying hardware IPV4-A Board is specifically designed for VoIP Internet telephony. The latest version IPV4-A is capable of fast voice data compression; run-time echo cancellation, and full-duplex transmission with de-suppression. It is designed with a wide range of circuit environments for most PSTN in any country or worldwide.
InterPhoneWork Frame Structure
For years of research in computer telephony and VoIP, has found that there is a need of an integrated Internet telephony system model that consists of a wide range of existing and future Internet telephony equipments. Therefore, we are trying to sculpt a complete and flexible Internet telephony framework known as InterPhoneWork.
Internet Telephony Network
Internet Telephony Network is based on InterPhoneWork Technology. There are over 30 gateways set up worldwide and expecting to expand to 100 in the short future. This complete Internet Telephony Network includes Embedded systems, Ethernet Phones, Gateways, and a complete Billing and Routing system.
IP2000 bridges the gap between PSTN and Internet when operated in a Phone-to-Phone mode, enabling customers to save enormous long distance charges.
IP2000 has the ability to communicate with IP Star and Inter Star. This unique feature offers ITSPs the benefit to turn IP2000 into a revenue centre.
Processor Pentium 166MHz or higher
Memory 32MB RAM or higher
Network 10/100BaseT LAN connection RJ-45 connector
Expansion Bus ISA slots for boards
Board IPV4-A (analog interface), IPV12-D (T1/E1 interface)
Capacity (per chassis) Up to 32 ports (analog interface)
Up to 96 ports (digital interface)
Operating System Windows NT Workstation 4.0
Build 1381/Service Pack 3
Remote Control(optional) SYMANTEC
Operation Modes Phone-to-Phone mode (ITSP/VPN)
Device-to-Phone mode (ITSP)
Multiple CODECs ITU G.723.1 6.3/5.3Kbps
G.728, G.722, G.729A/B
ITU G.723.1 Annex Comfort Noise Generation (CNG)
Voice Activity Detection (VAD)
PBX Integration C.O. line and extension access
Call progress analysis
Area code control
Advanced Algorithms Real-Time NT driver
Adaptive Jitter Buffer control
Mobile 44 7932 078 470
Fax: 44 207 681 3025
Mobile 44 7932 078 470
Fax: 44 207 681 302522nd June 2001 at 21:15 #20439pramGuest
I’m a student studying information science and technology and I need to know what voice over ip exactly is and what are its features.I would appreciate if anyone could help me out as I’m required to do a project
on this topic.
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