VoIP SIP Gateway
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-PBX
- 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, MEGACO and SIP control protocols
- 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 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.
TDM and SIP Trunking
AudioCodes’ Mediant E-SBCs and media gateways provide enterprises with a flexible and scalable solution for PSTN and SIP trunking connectivity.
- Hybrid TDM and SIP platforms provide flexible configurations for IP migration
- Successful homologations with PSTN networks worldwide
- Comprehensive support for SIP protocol variants ensuring seamless interoperability with softswitches and IP-PBXs
- Concurrent PSTN connection for Lifeline fallback and least cost routing
- Simplified centralized management
- Voice and service quality management integration
SIP Mediation and Interoperability
AudioCodes’ SBC’s SIP mediation capabilities translate between the different SIP variants implemented in the IP-PBX and the service provider network, allowing for a smooth and successful SIP trunk roll-out. AudioCodes products have proven interoperability with leading IP-PBX vendors and major SIP trunking service providers, ensuring that SIP trunks can be deployed quickly and with minimal effort.
Secured connection to SIP Trunking
AudioCodes’ E-SBC platforms offer enhanced IP security mechanisms to protect enterprises from malicious or fraudulent attacks such as:
- Denial of Service (DoS) protection
- NAT Traversal
- Topology hiding
- Malware and SPIT mitigation
- Dynamic Access Control List
- Advanced Call Admission Control (CAC)