SM7000 VoIP Gateway

SM7000 is a VoIP gateway that offers universal IP-PSTN switching, carrier grade reliability and high scalability. It supports the popular SIP and H.323 VoIP protocols, and enables service providers to quickly introduce revenue-generating VoIP services, such as class 4 service, callback, VoIP termination and others. Compared to other VoIP gateways, SM7000 offers more advanced features, such as callback and IVR over IP, and more compelling return on investment for service providers.

Key Features

• Deployable in SIP, H.323 or MGCP VoIP Networks
• Registration with Multiple Gatekeepers
• Support for Multiple RADIUS Servers
• Multilingual and Customizable IVR
• Support for IVR over IP
• Multiple Voice Codecs
• Route Fail-over Support
• SIP/H.323 Protocol Conversion
• Codec Translation
• Callback Support
  

Deployable in SIP, H.323 or MGCP VoIP Networks

SM7000 VoIP Gateway supports all major VoIP protocols, including H.323, SIP and MGCP and easily integrates into modern VoIP networks. The product also supports multiple PSTN protocols such as SS7, ISDN/PRI, CAS, GR-303, and MFC/R2 to ensure seamless connectivity with virtually any PSTN/SS7 network worldwide.

Registration with Multiple Gatekeepers

SM7000 VoIP Gateway can register with multiple third-party gatekeepers for flawless call routing. This unique gateway feature allows easy product integration with all major providers of call termination services. Gatekeepers can be specified separately for inbound and outbound routing purposes.

Support for Multiple RADIUS Servers

SM7000 VoIP Gateway can operate in complex billing environments with multiple RADIUS servers. For authentication and authorization purposes, SM7000 can work with a single RADIUS server; for accounting, however, SM7000 can communicate with multiple RADIUS servers simultaneously.

Multilingual and Customizable IVR

SM7000 VoIP Gateway offers enhanced IVR functionality with support for multiple languages and custom prompts. Such functionality enables providers to offer high level of service personalization by configuring SM7000 to interact with each subscriber at his/her own language.

Support for IVR over IP

SM7000 VoIP Gateway offers a unique IVR over IP functionality which enables it to encode and transport IVR messages over IP channels to gateways which don’t natively support IVR. Such product feature allows service providers to add IVR functionality to their existing VoIP infrastructure with low investment.

Multiple Voice Codecs

SM7000 VoIP Gateway supports multiple voice codecs, including G711, G723.1, G726, G729A, iLBC, SPEEXN, and GSM. All codecs can operate simultaneously on different ports of the gateway, thus ensuring interoperability with remote gateways supporting otherwise incompatible voice codecs.

Route Fail-over Support

SM7000 VoIP Gateway offers a mechanism to ensure high network availability. The product can be configured to periodically conduct L3, L4, and L7 remote service checks and re-route (fail-over) calls to alternative remote gateways if current terminals become unavailable.

SIP/H.323 Protocol Conversion

SM7000 VoIP Gateway ensures maximum interoperability with third party VoIP equipment through its protocol conversion capabilities. The product can translate signaling messages from SIP to H.323 and vice versa and thus bridge calls between VoIP equipment using incompatible protocols.

Codec Translation

SM7000 VoIP Gateway offers versatile bridging solution between VoIP terminals supporting incompatible voice codecs. The product can receive voice traffic from the origination gateway encoded in a particular format, decode that traffic and re-encode it in a format supported by the termination gateway.

Callback Support

SM7000 VoIP Gateway can be implemented in multiple callback scenarios requiring alternative callback initiation methods. The product can authenticate a subscriber by his/her PIN number send via web, SMS or email. Alternatively, it can initiate a callback based on the subscriber’s caller ID (ANI) or DNIS.

Key Features

• Multiple Routing Mechanisms
• Topology Hiding
• H.323/SIP Protocol Conversion
• Proxy Mode Support
• Routed Mode Support
• Routed Mode w/o H.245 Support
• Static Mode Support
• High Scalability
• ASR Route Switching
• Route Fail-over Support
• Universal Connectivity

Multiple Routing Mechanisms

uniSwitch Softswitch supports multiple routing mechanisms, including Least Cost Routing, ASR Routing, Priority Routing, Two-stage Routing, Preferred routing, Route Fail-Over and others. By providing multiple routing options, the VoIP softswitch enables providers to select the most profitable and high quality routes for each call and increase call completion rates, revenues and profits.

Topology Hiding

Using uniSwitch Softswitch, providers can securely separate their VoIP networks from the outside IP world. The VoIP softswitch can act as a traffic proxy between out-of-network VoIP terminals and in-network equipment thereby preventing outsiders from seeing the true topology of the protected network.

H.323/SIP Protocol Conversion

The protocol conversion functionality of uniSwitch Softswitch ensures maximum interoperability with third party VoIP equipment. The VoIP softswitch can translate signaling messages from SIP to H.323 and vice versa and thus bridge calls between VoIP equipment using incompatible protocols.

Proxy Mode Support

uniSwitch Softswitch can operate in proxy mode in which both signaling (RAS) and voice (RTP) traffic flow through the softswitch. Such bandwidth intensive mode is suitable for establishing connections to terminals behind NAT and to terminals that want to keep their identities private. In proxy mode the VoIP softswitch exercises full call control and can disconnect calls in progress if necessary.

Routed Mode Support

In routed mode, uniSwitch Softswitch requires less bandwidth. In such mode the VoIP softswitch performs direct control of signaling (RAS) messages only; voice (RTP) traffic is exchanged gateway-to-gateway. In routed mode the VoIP softswitch exercised substantial degree of call control and can disconnect calls in progress if necessary.

Routed Mode w/o H.245 Support

This operating mode of uniSwitch Softswitch mimics Routed Mode with the difference that H.245 call signaling is handled by the participating gateways, not the VoIP softswitch. Routed Mode w/o H.245 enables uniSwitch to reduce further bandwidth utilization while still preserving substantial degree of call control.

Static Mode Support

In static mode, both signaling (RAS) and voice (RTP) traffic are exchanged directly between participating gateways. In such mode uniSwitch Softswitch consumes least bandwidth but does not have control over calls in progress.

High Scalability

uniSwitch Softswitch is offered in standard and carrier editions to address the capacity needs of both emerging and established telecoms and service providers. Depending on the business model and infrastructure in which uniSwitch is utilized, the standard edition can support up to 500 concurrent calls, whereas the carrier edition can handle up to 1,000 concurrent calls.

ASR Route Switching

uniSwitch Softswtich offers ASR route switching capabilities to ensure higher call completion rates and reduced revenue loss. The VoIP softswitch monitors all end terminals in real-time and automatically disconnects those, whose ASR rates fall below critical values preset by system administrators.

Route Fail-over Support

uniSwitch Softswitch offers a mechanism to ensure high network availability. The VoIP softswitch can be configured to periodically conduct L3, L4, and L7 remote service checks and re-route (fail-over) calls to alternative remote terminals if existing ones become unavailable.

Universal VoIP Equipment Connectivity

uniSwitch Softswitch offers a number of features to ensure universal connectivity with other VoIP terminals. In particular, the VoIP softswitch can accept traffic from both registered and non-registered terminals and can terminate traffic to gatekeepers, gateways and softswitches.

uniSwitch Softswitch

uniSwitch is a VoIP Softswitch that offers flexible routing between VoIP networks, carrier grade reliability and high scalability. The product supports multiple routing methods, including Least Cost Routing, ASR Routing, and Priority Routing which enables providers to select the most profitable and high quality routes for each call, and increase call completion rates, revenues and profits. With the uniSwitch Softswitch, service providers benefit from improved call completion rates, higher revenues, less revenue leakage, and improved network security and availability.

Media5-Fone

The Media5-Fone is an application that runs on an Apple iPhone smartphone or iPod Touch. It is a SIP Client (softphone) that enables users to make and receive VoIP calls. VoIP calls are calls established over a Wi-Fi connection using the IP technology of the Media5-Fone. You can hence use your iPhone as an IP-PBX phone extension in your office or anywhere else in the world.

Universal SIP Interoperability and voice quality is achieved based on the award winning M5T SCE (SIP Client Engine) Mobility SDK, integrated and used around the world. Therefore, with the Media5-Fone application, your iPhone can be interconnected in a snap with all major SIP PBX providers; wether they are open source (such as Asterisk) or closed source (such as PBXnSIP).

The user manual describes how to install, configure, and use the Media5-Fone application. It does not describe how to operate the other functionalities of the device. Please refer to the device’s documentation.

Media5Boss-Branch

An industry-leading branch office solution specifically geared to support secure real-time communication scenarios like Voice-over-IP. The Media5Boss -Branch all-in-one systems enable companies to accomplish a branch consolidation of their IT infrastructure while assuring the quality of service, security and survivability of their communication infrastructure. One key element in this is the network convergence of data and voice networks. The Media5Boss -Branch Series fully supports this migration by providing Session Border Controller (SBC) functionalities.

Designed specifically for enterprise and hosted applications, the Media5Boss -Branch devices seamlessly integrate with your telecommunication equipment and your carrier connections while connecting you to any standard-based VoIP network via a broadband connection.

The Media5Boss -Branch differentiates itself from other branch office solutions in providing both Security and Survivability capabilities and because it can also act as a Session Border Controller for voice scenarios, thus controlling real-time traffic flows in and out of the branch office. These powerful features make it the premier single box branch connectivity solution for VoIP Enterprise or VoIP Centrex scenarios.

The Survivability and Call Admission Control capabilities, the availability of voice call routing and SBC functionalities, the ability to terminate SRTP calls, as well as the state-of-the-art QoS functionality make these systems a true one-box-solution for current generation branch offices or customer premise equipment.

Integrated Gateway Solution

The Media5Boss -Branch allows enterprises to lower communications costs over any IP link. They are multi-function devices combining VoIP IAD, Gateway, IP Router, SIP aware NAT/Firewall, VPN functionality, and QoS control in a secure and powerful platform.
This platform, featuring a broad offering of voice interfaces, including analog FXS, ISDN BRI, or T1/E1 PRI interfaces, provides an ideal solution for enterprise voice applications, hosted applications, or for connecting to a service provider’s broadband access.

High Performance, Great Flexibility

The Media5Boss -Branch is suitable as a universal network building block for medium-size offices. It helps to keep the costs down by providing an all-in-one, future proof solution. The performance of the Media5Boss -Branch allows connection for up to 150 users and providing them with all functionalities described in this document. It also features a redundancy solution that permits using a second Media5Boss -Branch in Hot-Failover mode for high availability

Comprehensive Real-time Communication Solution

The Internet is rapidly expanding, becoming increasingly content-rich and business-critical. The services offered will continue to grow both in terms of variety and quality, improving the ability to address new convergence scenarios derived by the combined usage of voice, text and video content and the integration of voice and data networks. Current data network platforms present severe limitations since they have not been specifically designed to satisfy the new requirements of convergence applications and services. This problem is further compounded when trying to bring such applications to remote branch offices, retail outlets, or even to broadband telecommuters. In that situation, a powerful real-time communications capable security, routing and session border controller solution is required in the branch office.

SIP Enabled

The Media5Boss-Convergence solutions are fully RFC 3261 compliant SIP Proxy – Registrar as well as B2BUA. As such, they can act in conjunction with a SIP PBX to provide Voice Routing, Survivability, Call Admission control, ENUM Lookup, Call Prioritization, Transcoding, as well as other features.

Enterprise Class Network Security Solution

The Media5Boss-Convergence solutions offer a state-of-the-art firewall with Layer 7 Traffic classification as well as VPN functionality. The Convergence solutions use encrypted voice signaling and management protocols. With two LANs and one DMZ interface, there is also the possibility to configure different security zones even on the LAN side. Contrary to many other branch office solutions, the Media5Boss-Convergence solutions can also act as a Session Border Controller for voice scenarios thus controlling real-time traffic flows in and out of the branch office. This makes it the first single box branch connectivity solution for VoIP Enterprise or VoIP Centrex scenarios.

Overview Enterprise Voice Route Helper is a Graphical User Interface (GUI) application that helps visualize, test, modify, archive, and share voice routing configuration information. Route Helper is an alternative to the Microsoft Management Console (MMC) snap-in for viewing and modifying Enterprise Voice number normalization rules, location profiles, voice policy, and routes. This webcast looks at creating and running configuration test cases to ensure the usability of routing data after making changes, but before committing them to a deployed system.