Rental/Sale Latest VoipSWITCH 2.0.0.879+All Modules+Training

Discussion in 'Ruby' started by Solution4voip Solution4voip, Jul 16, 2007.

  1. Dear Friends:

    Want to setup VOIP company, a business under your own brand name? We
    have complete solution to launche VOIP (VOice Over Internet Protocol)
    company. All support comes included.

    Softswitch is the main element of the platform, which merges the
    functionality of the following VOIP architecture’s elements.

    H323 switch

    H323 gatekeeper

    SIP Proxy

    SIP registrar


    Each of the described elements can operate simultaneously with the
    others. Moreover, the clients, regardless of the protocol, or the way
    they transfer connections, can connect between one another. This option
    allows connecting the networks, which because of the differences in
    implemented protocols or dialects inside the particular protocol, cannot
    directly transfer connection between one another. Implementing
    Solution4VOIP as a central traffic controller also introduces a number
    of additional management, supervision and network security
    facilitations.


    The main characteristics of the softswitch include:



    · Simultaneous and transparent support of SIP and H323 protocols
    (sip?h323 and h323?sip translator

    · Possibility of implementing various types of proxy (e.g. RTP-proxy or
    signaling proxy), possibility of choosing proxy for each prefix defined
    in dialing plan.

    · Advanced routing and rating system

    · Full internetworking with most commercially available switches,
    softswitches, session border controllers and VOIP gateways.

    · VOIP equipment support

    · NAT support both for SIP and h323 equipment

    · Calling to sip devices behind NAT (without the necessity of
    configuring NAT)

    · Calling among users registered to softswitch, support for dynamic IP
    addresses

    · Authentication of VOIP equipment

    o by IP address

    o by ANI

    o by h323id

    o by the pair of login/password (according to the SIP standard)

    · Flexible routing

    · Individual, integrated billing system

    · Managing pre-paid and post-paid accounts

    · Setting up users in the VSConfig program

    · Managing users, blocking, setting limits

    · Generating the groups of users and managing lots

    · Creating and managing tariffs, the possibility of attributing a tariff
    to an individual user

    · Data stored in the MSSQL or MySQL database

    · Graphic management interface (presentation of the statistical data,
    billing information, managing clients’ accounts, generating PIN,
    managing the tariffs, dialing plan and others)

    · Graphic interface presenting the current traffic in the real time,
    number of the logged in clients, with the division into different types
    of services, presentation of logs and others

    · Web interface for clients – presentation of the connections history,
    possibility of exporting to the file, presentation of the current
    account status, possibility of making payments online and others

    · Easy to set up architecture

    · Automatic software re-start facilities in case of system failure

    · Scalability for new telecommunication services by enabling additional
    modules



    Advantages of managing the system:


    · Simplify the management processes and network configuration changes of
    VoIP equipment

    · Unify equipment supporting different protocols (or dialects of one
    protocol)

    · Manage concentration and routing processes of VoIP traffic

    · Centralize authorization and billing tasks of VoIP calls in one point

    · Hide the network structure from third parties, if necessary

    · Utilize possibility of implementing value-added services such as:
    calling card and DID calling card system, IPPBX, SMS/ANI/PIN/DID
    callback system.



    STANDARD APPLICATIONS


    Central point of your VOIP network



    Main benefits:


    Management of authorization rules of VoIP-gateways

    Setting up call routing rules

    Provisioning of compatibility for H323 and SIP- equipment of various
    vendors

    Security and load planning of VoIP-traffic by using optional
    RTP-proxying

    Access to the statistical data (ASR, PDD and others)

    Transparent interface of the billing system



    Network security



    When using RTP-proxying SoftSwitch provides a single entry point for
    VoIP traffic.Both for clients and carriers there is only one IP address
    available.

    Integration of equipment with support of different protocols

    One of the most important features of RSF1000 is its ability to support
    widely accepted signaling IP-protocols - SIP and H323. The system
    provides transparent converging of one protocol into another, thus
    allowing performing calls from one type of equipment to another.



    SCALABILITY


    Through launching subsequent modules, it is very convenient for a
    provider to extend the range of services offered. Available modules:

    IVR for calling cards

    Web/SMS/ANI callback (with IVR)

    Reseller’s module

    Online shop

    CallShop



    SPECIFICATIONS


    Supported protocols


    1 H.323 v.2 (H.245 v7, H225 v4) with/without FAST START

    2 SIP (RFC 3261)

    3 proxying of RTP/RTCP streams

    4 Signalling proxy

    5 Support of T38 (SIP, H323)

    6 Transparent conversion of SIP to H323 and vice versa



    Support of the Devices Behind the NAT


    1 SIP-devices

    2 H323-devices



    Authentication


    1 by IP address – SIP and H323

    2 by H323ID – h323 terminals/gateways

    3 by ANI (calling party number) – SIP and H323

    4 by login and password- SIP equipment

    5 by login and password – HearLink pc to phone/web to phone dialer
    (included in the package)

    6 gatekeeper registration based on aliases



    Intelligent routing


    1 based on prefixes (the possibility of defining prefixes
    differentiating individual users)

    2 based on accessibility of the VOIP gateway

    3 based on priorities when choosing a gateway

    4 depending on available voice codecs

    5 depending on prefixes specified in the tariff of an individual client


    Phone Numbers Translation


    1 Deletion of the set number of digits from the called party number

    2 Addition of the set number of digits to the called party number

    3 Deletion of the set number of digits from the caller number

    4 Addition of the set number of digits to the caller number

    5 Virtual prefixes (for differentiation of the dialing plans)



    Information for the Billing System


    1 Real-time, built in billing system

    2 Storage in SQL database (MSSQL or MYSQL)

    3 pre-paid and post-paid accounts

    4 Payments history

    5 CDR – examining the logs of the calls carried out from the VSCConfig
    level, possibility of filtering data according to the set parameters,
    possibility of exporting data to the file (html, excel, txt, or csv
    type), presenting the CDR on the WWW pages available for clients



    System Management and Control Features


    1 Graphic User Interface for managing the overall functionality of the
    system

    2 Visual presentation of current connections along with the information
    on their status

    3 The number of statistical data presenting the information on the
    traffic intensity with its various parameters e.g. ASR, PDD. Possibility
    of limiting the number of data presented by using available filters e.g.
    only incoming traffic from the particular client, traffic directed to
    the particular gateway, or prefix etc.

    4 Visual presentation of logged in clients and their current status,
    with the division into types of services e.g. gatekeeper users, SIP
    users, pc2phone, callback.



    Operating Systems

    1 Windows 2000, 2003, XP


    ------------------------------------------------------------ -----

    Contact us if you are interested.


    Thank you,
    Solution4voip.com
    VOIP Solution Provider
    Solution for Voice Over Internet Protocol (VOIP)

    Email: salesATsolution4voipDOTcom
    MSN: SalesATsolution4voipDOTcom, SupportATsolution4voipDOTcom
    Phone: +1800 7816433

    --
    Posted via http://www.ruby-forum.com/.
     
    Solution4voip Solution4voip, Jul 16, 2007
    #1
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