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2FXS 1FXO SIP SPE VoIP Gateway

Product ID: SPE-2.1 pro

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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 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 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).
Specifications
  • VoIP Protocols
    • SIP (RFC 3261~3266 and various SIP draft extensions)
  • SDP(RFC2327)
  • RTP/RTCP(RFC 3550, 3551) Call Features
    • Call Transfer (Attended)
  • Call Forward (Busy / No Answer /Unconditional)
  • Call Hold / Retrieve
  • Call Waiting
  • Caller ID Display
  • Caller ID Blocking
  • Anonymous Call Blocking
  • Hook Flash FAX Support
    • G.711 pass- through
  • T.38 Codec
    • G.711μ-law
  • G711a-law
  • G.729, G.729a/b
  • G.723 DTMF
    • In-band DTMF
  • Out-of-band DTMF(RFC 2833)
Features
  • Dial Method
    • Direct IP call without SIP proxy
  • Dial number via SIP server
  • Dial URI via SIP server
  • Flexiable Dial Plan Router
    • Virtual Server
  • NAT
  • DHCP Server
  • DMZ Voice Quality
    • VAD (Voice Activity Detection)
  • CNG (Comfort Noise Generation)
  • AEC (Acoustic Echo Cancellation)
  • AGC
  • Packet Loss Concealment
  • Each Line G.168 up to 128ms
  • Dynamic Jitter buffer
  • Bell 202 and V.23 Caller ID TCP/IP
    • IP/TCP/UDP/DHCP/RTP/RTCP
  • ICMP/HTTP/SNTP/TFTP/DNS Configuration & Management
    • Web browser
  • Touch-tone telephone keypad configuration with voice prompt
  • Auto-provisioning scheme with Aegis QuickStep
  • SpeedVoIP IntelliManage? Reverse Control even residing after NAT/Firewall
  • Periodic "HeartBeat" Monitor
Certificate
  • CE
  • FCC
  • UL
  • RoHS.

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