VOIPSWITCH OVERVIEW
Voipswitch is a software platform allowing for rapid voip services roll-out. It contains all necessary elements required in successful implementation of various VoIP services.
| VOIPSWITCH FEATURES: |
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A highly scalable softswitch with integrated billing. |
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Class 5 features included. |
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A Calling card platform with IP IVR including pin or pinless scenarios. |
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A Callback solution supporting all types of triggering methods, e.g. SMS, missed call, web call back. |
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Web callshop interface. |
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Web self-care portal for end users and support for Online payments. |
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SMS features for both wholesale and retail services, using http, SIP and SMPP protocols. |
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Mobile SIP for Symbian, Windows Mobile and Iphone users; all features are integrated with VoIP tunneling. |
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Windows communicator with Instant Messaging, SMS, voicemail, sip calling and many other modules. |
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Multi-tenant, multi-user IP PBX platform. |
| VOIPSWITCH CLIENTELE : |
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Wholesale termination carriers. |
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Retail internet telephone service providers. |
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Internet providers including WIFI and WIMAX operators extending their offer by adding VoIP. |
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Cable TV networks and incubent telecoms. |
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GSM providers. |
| VOIPSWITCH EXPERTISE: |
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Actively involved in the VoIP arena since 2001. |
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In excess of 1500 Voipswitch solutions deployed worldwide. |
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Award winning softswitch solution, Best product of the year 2008 and Next Generation Network Leader 2009. |
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Voipswitch has a presence at all important fairs and VoIP conferences worldwide. |
Our customers can make money on whole range of services starting with wholesale voice and sms termination, through calling cards (phone to phone) and all types of callback (ANI, PIN, DID, SMS, Web), to the services offering full replacement of traditional phones, namely broadband phone calling, which enables users to benefit from 5 class services (voicemail, own phne numbers, additional features).
In addition Internet Telephony Service Providers (ITSP) can enhance their offer by adding softphone/communicator with sip, instant messenger and other cutting edge functionality. Another popular group of phone services are callshops which are also fully supported by our software.
All services can be offered through resellers who can manage their endusers accounts, rates sheets, see reports, active calls and more via web interface.
The system is multilayered, with the core which is the border session manager (softswitch) integrated with own, built-in, billing system. The core is responsible for switching the traffic, accepting incoming connections in different protocols, authorization, billing, reconcilation of different protocols or dialects and proxying the packets.
The higher layers consists of the applications extending the core of additional functionality like callback, Interactive Voice Response System, Callshop engine, Resellers module and others.
In addition the platform offers variety of tools for managent and reporting such as VSM, Web Config, Event Manager and others.
Another group of software are applications designated for endusers to which belong VSPortal - endusers web interface, softphones - Vippie!, SIPLink, Vippie mobile and also Callshop web interface and Online shop.
The softswitch part is the main element of the platform, which merges the functionality of the following
| VOIP ARCHITECTURE ELEMENTS : |
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SIP registrar. |
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SIP proxy). |
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SIP proxy). |
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H323 gatekeeper. |
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SMS gateway. |
| SUPPORTED PROTOCOLS: |
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H323 v.2 (H245 v7, H225 v4) with/without “fast start”. |
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SIP (RFC 3261). |
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SMS through SIP, HTTP and SMPP. |
| CHARACTERSTICS OF VOIPSWITCH : |
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Simultaneous and transparent support of SIP and H323 protocols (SIP<->H323 translator).
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Various types of proxy methods e.g. full proxy (with RTP-proxy), signaling proxy and other options, possibility of selecting a proxy method per destination, route or per client
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Full interoperability with industry standards compatible VoIP equipment (gateways, switches, ATAs, terminals) |
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Bidirectional NAT supports both for SIP and H323 equipment. |
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Advanced routing system (support for internal virtual prefixes that allows the creation of separate dialing plans for different groups of customer accounts)
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Routing based on prefixes, priorities per routes, depending on allowed voice Codecs per destination.
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Support for failover (rerouting), configurable end reasons initiating fail over, support for priorities
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Load sharing support – Advanced algorithm taking care of traffic being evenly distributed according to defined percentages for multiple routes
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Load sharing support – Advanced algorithm taking care of traffic being evenly distributed according to defined percentages for multiple routes
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Internal numbering plans support
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| VARIOUS AUTHETICATION METHODS: |
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By IP address. |
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By caller ID (ANI). |
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By H323 ID. |
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By SIP credentials. |
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SIP. |
| FLEXIBLE METHODS FOR ALL CALL SETUP DATA MODIFICATIONS (for clients and/or for destination in the dialing plan): |
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Modifying dialed number, adding prefixes/suffixes, wild cards, max/min number length. |
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Modifying caller ID/SIP display – Adding prefixes/suffixes, wild cards, max/min number length |
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Defining allowed and/or primary Codecs for clients and for terminators. |
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Codecs auto negotiation. |
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Import/Export accounts and dialing plan from/to excel or TXT file |
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Settings stored in the MySQL database.. |
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Scalability is supported due to a cluster configuration, where multipleVoipSwitch servers run connected to each others in what is known as “cluster”, sharing the same SQL database server, thus increasing performance by dividing the traffic among the multiple servers while retaining a central point of management with one main IP address for clients (load balancing). |
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Scalability is supported due to a cluster configuration, where multipleVoipSwitch servers run connected to each others in what is known as
Redundancy support for seamless traffic handover in case of the main server failure, the service allows for controlling availability of particular ports (for example SIP, H323 listeners) real-time SQL data backup procedure.
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| TELEPHONY FEATURES : |
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Hold. |
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Music on hold. |
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Do not disturb. |
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Follow me/Find me (based on caller ID of incoming call), sequential or ring to all. |
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Voicemail boxes with personalized voice greetings for different caller IDs. |
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Call transfer: blind and attended. |
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Hunt/Ring groups. |
| UNIFIED MESSAGING: |
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Voicemail to email with attachment (Mp3). |
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Voicemail notification to SMS or email. |
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Voicemail transcription to SMS. |
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SMS forwarding (e.g. internal SIP SMS forwarding to external GSM numbers). |
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