The new Mediant 1000B chassis, based on the incumbent Mediant 1000 chassis is designed to support the Advanced Mezzanine Card (AMC) or AdvancedMC form-factor modules. It provides eight AMC slots (for housing single and mid-sized AMC modules) on its rear panel.
The Mediant 1000B supports both VoIP Gateway and MSBG data routing functionalities, by hosting either the CMX or CRMX module, respectively. The Mediant 1000B also provides support for the regular Mediant 1000 telephony modules (i.e., FXS, FXO, and digital PSTN).
The Mediant 1000B chassis supports (optional) the latest OSN server platform - OSN3. The OSN3 provides AMC-based modules, offering high performance CPUs targeted for applications requiring high performance processors. The OSN3 can also be provided with dual SATA hard-disk drives (HDD).
The AudioCodes Mediant 800 Enterprise Session Border Controller (E-SBC) and Media Gateway offers a complete connectivity solution for small-to-medium sized enterprises. Supporting up to 60 voice channels in a 1U platform, the Mediant 800 provides versatile connectivity between TDM and VoIP networks.
The Mediant 800 connects IP-PBXs to any SIP trunking service provider, scaling to 250 concurrent sessions. It offers superior performance in connecting any SIP to SIP environment, legacy TDM-based PBX systems to IP networks, and IP-PBXs to the PSTN.
The HT812 delivers powerful VoIP technology and routing capabilities to home and office environments, and allows users to successfully connect their analog devices to a manageable and powerful VoIP network. Built using Grandstream's market-leading SIP ATA/gateway technology, with millions of units successfully deployed worldwide, this powerful ATA features exceptional voice quality in various applications and environments. The HT812 comes with 2 easy-to-use FXS ports, an integrated Gigabit NAT router, state-of-the-art encryption with a unique security certificate per unit, automated provisioning for volume deployment and device management and outstanding network performance.
Supports 2 SIP profiles through 2 FXS ports and dual Gigabit ports
Includes a built-in NAT router which can handle routing speeds up to 100MBps
TLS and SRTP security encryption technology to protect calls and accounts
Automated provisioning options include TR-069 and XML config files
Supports 3-way voice conferencing
Failover SIP server automatically switches to secondary server if main server loses connection
Supports T.38 Fax for creating Fax-over-IP
Supports a wide range of caller ID formats
Use with Grandstream's UCM series of IP PBXs for Zero Configuration provisioning
The GXW4500 series are E1/T1 Digital VoIP Gateways that allow digital PSTN and ISDN trunks to be integrated with VoIP networks. By connecting the GXW4500 series with a VoIP network and a traditional PBX or E1/ T1 provider, businesses can drastically increase the amount of PSTN/ISDN trunks integrated with their VoIP network. The GXW4500 series offers three models that provide 1, 2 or 4 E1/T1/J1 spans and support 30, 60 or 120 concurrent calls to cater to the VoIP needs of large and medium sized enterprises.
Dual Gigabit autosensing RJ45 network ports with integrated NAT router
TLS and SRTP security encryption technology to protect calls and accounts
Supports a widerange of voice codecs, including G.722, G.729, iLBC, and more
The Cisco® ATA 191 Analog Telephone Adapter turns traditional telephone, fax, and overhead paging communications devices into IP devices for greater cost-effectiveness. Customers can take advantage of IP telephony applications by connecting their analog devices to Cisco analog telephone adapters.
The Cisco ATA 191 Analog Telephone Adapter is the preferred solution to address the needs of customers who connect to enterprise networks, small offices, or the emerging unified communications as a service from the cloud. It has two standard FXS ports, which can be configured independently as two Session Initiation Protocol (SIP) registrations. With the ATA 191, customers can protect and extend their existing investment in analog systems, as well as smooth their migration to pure voice over IP in an affordable and reliable way.
The AudioCodes MediaPack™ Series of analog VoIP gateways are cost-effective, best-of-breed technology products. These stand-alone gateways provide superior voice technology for connecting legacy telephones, fax machines and PBX systems with IP-based telephony networks, as well as for integration with IP PBX systems.
Fully interoperable with leading softswitches and SIP servers, MediaPack gateways are ideal for commercial VoIP deployment due to their mature and field-proven voice and fax technology. Their rich feature set allows integration with a wide range of carrier and enterprise network applications.
2 to 24 Analog Ports | Zero-Touch Provisioning | Standalone Survivability | T.38 Fax Compliant
Wide versatility Provides voice, fax and modem support
High resiliency Standalone survivability (SAS) and fallback to PSTN for E911 (emergency number PSTN breakthrough) or upon network/power failure
Broad interoperability Proven integration with leading PBXs, IP-PBXs and softswitches
Superior quality Toll quality voice compression
Enhanced capabilities Includes MWI, long haul, metering tones generation and caller ID
AudioCodes' MediaPack 1xx series of analog VoIP gateways offer service providers and enterprises superior voice technology for connecting legacy telephones, fax machines and PBX systems with IP telephony networks and IP-PBX systems.
The MediaPack 1xx gateways are fully interoperable with leading softswitches and SIP servers.
Benefits
Leverage investment in legacy analog telephone, modem, and fax systems - easing VoIP migration
Secured zero-touch provisioning, useful for large-scale deployments
Standalone Survivability (SAS) keeps your business running in the event of a network failure
The Cisco® VG350, VG320, VG310, VG204XM, and VG202XM Analog Voice Gateways allow you to use your IP telephony solution with traditional analog devices while taking advantage of the productivity afforded by IP infrastructure.
Unified communications enables organizations to collaborate more effectively and streamline business processes. Reach the right resource the first time. Improve productivity and profitability. Even in an IP world, however, many organizations still use analog devices such as phones, faxes, and modems, and want or need to continue to after migration to IP telephony. Important use cases for analog gateways include:
Budget constraints during migration to unified communications: You don't have to replace your entire phone system as you move to unified communications. Cisco VG Series Gateways help you migrate at your own pace and budget: you can enjoy the benefits of unified communications and still use your existing analog devices.
Meeting regulatory requirements: Many countries or industry markets require analog devices in certain use cases. Using Cisco VG Series, these organizations can still meet regulatory requirements but also have the benefits of unified communications.
Environments that require phone service but do not have power and require long loop lengths: Examples include ranger stations, phones along railroad tracks, and some military deployments such as remote outposts.
Harsh environments: Industries such as mining and manufacturing have harsh environments in which analog phones often have better endurance.
Requirements for lighter IT infrastructure: Analog phones need fewer switch ports and have no power requirements.
The AudioCodes Mediant 800 enterprise session border controller (E-SBC) and media gateway offers a complete connectivity solution for small-to-medium sized enterprises. Supporting up to 124 voice channelsin a 1U platform, the Mediant 800 provides versatile connectivity between TDM and VoIP networks.
The Mediant 800 connects IP-PBXs to any SIP trunking service provider, scaling to 400 concurrent sessions. It offers superior performance in connecting any SIP to SIP environment, legacy TDM-based PBX systems to IP networks, and IP-PBXs to the PSTN.
Benefits
Broad support for qualified SIP trunks, SIP platforms and IP cloud services
Hybrid platform lowers CAPEX and reduces space and power footprints
Simplified integrated management reduces OPEX
Single and managed point of demarcation for VoIP networks
Optional integrated server for hosting value-added applications
The Cisco ATA 192 Multiplatform Analog Telephone Adapter turns traditional telephone, fax, and overhead paging communications devices into IP devices for greater cost-effectiveness. Customers can take advantage of IP telephony applications by connecting their analog devices to Cisco analog telephone adapters.
The ATA 192 is the preferred solution to address the needs of customers who connect to enterprise networks, small offices, or unified communications as a service from the cloud. It has two standard FXS ports, which can be configured independently as two Session Initiation Protocol (SIP) registrations. It also has two 100BASE-T ports with an integrated high-performance router to extend local network connectivity. With the ATA 192, customers can protect and extend their existing investment in analog systems, as well as smooth their migration to pure voice over IP in a more affordable and reliable way.
The ATA 192 is designed to work with third-party call control systems and does not work with Cisco call control systems.
The HT818 is a powerful 8-port VoIP gateway with 8 FXS ports and an integrated Gigabit NAT router. Built for users looking for a strong analog-to-VoIP converter, it features Grandstream's market-leading SIP ATA/gateway technology with millions of units successfully deployed worldwide. This powerful gateway carries exceptional voice quality in various application environments, strong encryption with unique security certificate per unit, automated provisioning for volume deployment and device management, and outstanding network performance for enterprise use.
Supports 2 SIP profiles and 8 FXS ports
Strong AES encryption with security certificate per unit
Automated & secure provisioning options using TR069
MediaPacks are well suited for commercial VoIP deployment because of their mature and field-proven voice and fax technology. Their rich feature set allows integration with a wide range of Carriers and Enterprise network applications. MediaPack gateways are used by Carriers and Service Providers in Access networks for connecting Multi-Tenant Units (MTU), IP Centrex subscribers, payphones and rural users over various wireless and satellite links. Enterprises use MediaPack gateways to connect their legacy PBX systems over an IP infrastructure. In addition, in IP Centrex and central IP-PBX applications, the MediaPack increases the remote location availability and provides Stand Alone Survivability (SAS) when there is no IP connection between branch locations and the central SIP servers, SIP Proxy or central IP-PBX.
The AudioCodes Mediant 800 enterprise session border controller (E-SBC) and media gateway offers a complete connectivity solution for small-to-medium sized enterprises.
Scaling up to 400 concurrent sessions, the Mediant 800 connects IP-PBXs to any SIP trunking service provider and offers superior performance in connecting any SIP to SIP environment.
In addition, the Mediant 800 supports up to 124 voice channels in a 1U platform to enable versatile connectivity between TDM and VoIP networks, such as connecting legacy TDM PBX systems to IP networks and IP-PBXs to the PSTN.
Comprehensive interoperability
Proven interoperability with SIP trunks, SIP platforms and IP cloud services
Hybrid functionalit
True hybrid SBC and gateway platform for gradual migration, low CAPEX and reduced space and power footprints
Enhanced security
Robust perimeter defense against cyber, DoS and DDoS attacks, as well as eavesdropping, fraud and service theft
Superior voice quality
Advanced capabilities for optimizing and monitoring voice service quality
High resiliency
High availability using 1+1 redundancy, local branch survivability and PSTN fallback
The AudioCodes Mediant 4000 is a Session Border Controller (SBC) designed for deployment in large organizations and as an access SBC for service providers. Supporting up to 5000 concurrent sessions, the Mediant 4000 supports extensive SIP connectivity with wide-ranging interoperability, enhanced perimeter defense against DoS attacks, and advanced voice quality monitoring.
MediaPacks are well suited for commercial VoIP deployment because of their mature and field-proven voice and fax technology. Their rich feature set allows integration with a wide range of Carriers and Enterprise network applications. MediaPack gateways are used by Carriers and Service Providers in Access networks for connecting Multi-Tenant Units (MTU), IP Centrex subscribers, payphones and rural users over various wireless and satellite links. Enterprises use MediaPack gateways to connect their legacy PBX systems over an IP infrastructure. In addition, in IP Centrex and central IP-PBX applications, the MediaPack increases the remote location availability and provides Stand Alone Survivability (SAS) when there is no IP connection between branch locations and the central SIP servers, SIP Proxy or central IP-PBX.
The GXW4216/24/32/48 is a next generation high performance high-density analog VoIP gateway that is fully compliant with SIP standard and interoperable with various VoIP systems, analog PBX and phones on the the market. It features multiple FXS analog telephone ports, superb voice quality, rich telephony functionalities, easy provisioning, flexible dialing plans, advanced security protection, and strong performance in handling high volume voice calls. The GXW42XX series gateway offers small and medium businesses a cost-effective hybrid IP and analog telephone system that allows them to enjoy the benefits of VoIP communications while preserving investment on existing analog phones, Fax machines and legacy PBX systems.
The HT801 delivers powerful VoIP technology and routing capabilities to home and office environments, and allows users to successfully connect their analog devices to a manageable and powerful VoIP network. Built upon Grandstream's market-leading SIP ATA/gateway technology, with millions of units successfully deployed worldwide, this powerful ATA features exceptional voice quality in various applications and environments. The HT801 comes with 1 easy-to-use FXS ports, state-of-the-art encryption with a unique security certificate per unit, automated provisioning for volume deployment and device management and outstanding network performance.
Supports 1 SIP profile through a single FXS port and a single 10/100Mbps port
TLS and SRTP security encryption technology to protect calls and accounts
Automated provisioning options include TR-069 and XML config files
Supports 3-way voice conferencing
Failover SIP server automatically switches to secondary server if main server loses connection
Supports T.38 Fax for creating Fax-over-IP
Supports a wide range of caller ID formats
Use with Grandstream's UCM series of IP PBXs for Zero Configuration provisioning
Supports advanced telephony features, including call transfer, call forward, call-waiting, do not disturb, message waiting indication, multilanguage prompts, flexible dial plan and more
