Q-1FXS-SIP-PP & Q-2FXS-SIP-PP™

Proven effective, low-cost VoIP gateway for IP-based Voice and Real-Time Fax Communication

Designed for use with Ecocarrier's service platform to do unblockable VoIP operation by running proprietary protocol

Ideal for use in Call Shops, Small Office and Home Office

Applications

For use as a standalone VoIP gateway for a telephone service provision business in a Call Shop operation

• connect individual regular analog telephones into Q-1FXS-SIP-PP or Q-2FXS-SIP-PP and use it to provide prepaid or post-paid Long Distance Telephone Call service to customers for 1 or 2 calls simultaneously in a Call Shop operation
• all Long Distance Calls are automatically placed through Ecocarrier service platform to call destination telephone numbers at extremely low rates - providing the Call Shop operator with significant competitive advantage
• small offices of an enterprise and home offices of employees of an enterprise may use  Q-1FXS-SIP-PP  or Q-2FXS-SIP-PP for a low-cost way of equipping the offices for IP telephony
• creation of a virtual private telephone network that is implemented with facility for inter-company dialing and call traffic data record and accounting that enhance the business operation of the enterprise

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(1/2/4/8/16/24) 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.723, G.729AB and G.711 (A law and U-law), FAX and Modem pass-through
• Supports PSTN-Relay in case of power failure and external control
• Supports DIGEST authentication (MD5 only)
• 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.168 128ms
• 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)

Software Specification