Product Description

The SPE Gateway is capable of delivering an ultra-affordability solution for commercial and residential application worldwide, and meets international branches, SOHO, and personal needs. Being composed of a voice coprocessor, network processor and voice codec, SPE Gateway is a Handset-to-Ethernet adaptor that can convert human voice from an analog signal to a digital signal and transmit it out over Internet or any IP-based network. It is a low cost, feature-rich SIP customer premise equipment (CPE) device that supports router, VoIP, FoIP and remote software upgrading functionalities and can be installed seamlessly in any SIP based telephony system. It offers advanced VoIP and Peer-to-Peer technologies that substantially reduce the cost of long distance/International calls and to cost-effectively scale their telephone systems as their communication needs grow. Available in *-, *-, *-, **- and **-port models, the SPE gateway connects directly to phones, fax machines, key systems, PSTN lines, or a PBX. Our SPE gateway integrates voice and fax communications into your data network providing distributed IP telephony and toll bypass savings to remote offices of multi-location businesses. Full Feature Voice/Fax-over-IP Ultra-Compact and Lightweight Universal Plug and Dial Extremely Cost Affordable Product Key Features Supports TCP/UDP/IP, RTP/RTCP, HTTP, ICMP, ARP/RARP, DNS, DHCP (both client and server), PPPoE, TFTP etc Built-in router, NAT and Gateway Supports SIP Multiple(1/2/4/8/*6/*4) FXS and necessary FXO ports with independent telephone numbers Simultaneous signaling and media encrypting and mangling Free access to customer network Intelligent voice routing and discovery Interoperable with various market-leading third party Softswitch or Sip Server Supports popular voice-coders including G.**3, G.**9AB and G.**1 (A law and U-law), FAX and Modem pass-through Supports PSTN-Relay in case of power failure and external control Supports standard voice features such as Caller Id, Consultation Hold, Call Waiting, Blind Call Transfer, Attended Call Transfer, Call Forward, Mid-Call DTMF, Flash, Three-way Conference Supports Silence Suppression, VAD (Voice Activity Detection), CNG (Comfort Noise Generation), AGC and Line Echo Cancellation G.**8 **8ms Supports DIGEST authentication (MD5 only) Supports layer*3 (DiffServ, ToS) QoS Supports automated NAT traversal without manual manipulation of firewall/NAT Supports remote automated provisioning and software upgrade even through firewall/NAT to enable �zero configuration� and �plug-and-dial� for end users Supports device configuration via built-in IVR by dialpad, Web browser or central configuration file Intelligent powerful proprietary AVMP management scheme Standards-based implementation (ITU-T, IETF compliant).