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AudioCodes’ MediaPack 1xx series of Analog VoIP Gateways are cost-effective, stand-alone VoIP gateways that provide superior voice technology for connecting legacy telephones, fax machines and PBX systems with IP telephony networks and IP-based PBX systems. The MediaPack 1xx series gateways support a wide variety of service provider and enterprise applications:- Service providers can use MediaPack gateways to connect Multi-Tenant Units (MTUs), IP Centrex subscribers, payphones, and rural users over wireless and satellite links- Enterprises can use MediaPack gateways to connect their legacy PBX systems over an IP infrastructure. In addition, in IP Centrex and central IP-PBX applications, MediaPack enhances remote location availability and provides Stand Alone Survivability (SAS) when there is no IP connection between branch locations and a central SIP server, SIP proxy or central IP-PBXThe MediaPack 1xx gateways are fully interoperable with leading softswitches, SIP servers and H.323 systems.Features- Support for 2 to 24 analog ports- T.38 fax compliant- Rich subscriber feature set: 3-way conference with local mixing, call pickup, hunt groups, call forwarding, call hold and call transfer- Echo cancellation, jitter buffer, voice activity detection (VAD) and comfort noise generation (CNG)- Complies with MGCP, H.323v4 and SIP control protocolsBenefits- Leverage investment in existing analog telephone, modem, and fax systems – easing VoIP migration- Lifeline for fallback to PSTN for E911 (Emergency number PSTN breakthrough) or upon network/power failure (FXO and/or FXS configurations)- Standalone Survivability (SAS) keeps your business running in the event of a network failure

The AudioCodes MediaPack 5xx series of analog VoIP gateways provide service providers and enterprises with superior voice technology for connecting legacy telephones, fax machines and PBX systems with IP telephony networks, IP-PBX systems and unified communications solutions.MediaPack 5xx gateways offer comprehensive SIP interoperability with leading vendors, delivering secure and reliable communications for enterprises and branch office locations.The MediaPack 5xx series is compatible with the Microsoft SIP Gateway to enable the direct integration of analog devices with Microsoft Teams Phone.Comprehensive interoperabilityProven interoperability with SIP trunks, SIP platforms and IP cloud servicesSuperior voice qualityAdvanced capabilities for optimizing and monitoring voice service qualityEnhanced capabilitiesIncludes MWI, long haul, metering tones generation and caller IDAll-in-one optionOption to add data routing capabilities within the same deviceHigh resiliencyStandalone survivability (SAS) and fallback to PSTN for E911 (emergency number PSTN breakthrough) or upon network/power failure

SFB 450HD IP-PHONE POE GBE BLACK 2 ETHERNET 10/100/1000 PORTS, 800X480 5 COLOR T UC450HDEGNS Audiocodes Limited UC450HDEGNS Audiocodes Limited

MEDIANT 9030 SBC, DC POWER M9K30/DC Audiocodes Limited M9K30/DC Audiocodes Limited

AudioCodes Mediant 9080 VoIP Gateway - 10 Gigabit Ethernet, Gigabit Ethernet

The AudioCodes Mediant 1000 Enterprise Session Border Controller (E-SBC) and Media Gateway is designed to provide a complete connectivity solution for small-to-medium sized enterprises. Supporting up to 192 concurrent voice sessions in a 1U modular platform, the Mediant 1000 provides versatile connectivity between TDM and VoIP networks.The modular Mediant 1000 connects IP-PBXs to any SIP trunking service provider, scaling to 150 concurrent sessions. It offers superior performance in connecting any SIP to SIP environments, legacy TDM-based PBX systems to IP networks, and IP-PBXs to the PSTN.Benefits- Support a wide variety of qualified SIP Trunks, SIP Platforms and IP Cloud services- The “all-in-one” VoIP Media Gateway and E-SBC lowers CAPEX and reduces space and power footprints- Simplified integrated management reduces OPEX- A highly integrated device for secured SIP Trunking and PSTN access, forming a single and managed point of demarcation for VoIP networks- Scalable “pay-as-you-grow” modular architectureFeatures- Rich and powerful SIP normalization and routing mechanism for seamless interoperability- Support for analog and digital TDM interfaces- Hybrid SBC enables seamless migration and PSTN fallback- Enhanced perimeter defense against DoS attacks- Advanced monitoring tools help analyze and optimize VoIP service quality-of-experience- Embedded DSP performs media handling (codec transcoding, DTMF, fax, call progress announcements, etc.)- E1/T1/J1 bypass and dual power supplies for high-availability