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Symetrix 2 LINE VOIP INTERFAC SIP Based Hardware Plug In Card Data Sheet

Symetrix 2 LINE VOIP INTERFAC SIP Based Hardware Plug In Card Data Sheet

Symetrix

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Symetrix 2 LINE VOIP INTERFAC SIP Based Hardware Plug In Card Data Sheet

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Natural Sounding Wide-band VoIP Audio

The Symetrix 2 Line VoIP Interface Card is a SIP-based hardware plug-in for Symetrix Radius and Edge DSPs. The card supports both
narrow and wide-band audio codecs and is capable of a broad range of telephony functions including dial, hold, resume, transfer, do not
disturb, and conference. Independent SIP registrations are provided for two simultaneous calls. The 2 Line VoIP Interface Card is well suited
for applications in conferencing, paging, remote monitoring, and broadcast. 

Open Standards Interoperability

Designed around open SIP standards, the card has been
validated with Cisco, Avaya, and Asterisk SIP-compatible call
management platforms. The card delivers high-definition voice
quality and a superior unified communications experience, while
leveraging and complementing existing telephony investments. 

Best-In-Class Deployment and Administration

Deployment and administration are intuitive and flexible.
Configuration can be managed from both the AV and VoIP LANs,
giving AV integrators and VoIP specialists easy access for initial
setup and maintenance.

Multiple Control Options

SymVue application builder and extensive third-party control support allow for intuitive end-user operation, while reducing the amount of time needed
to develop application-specific user interfaces.

Product Highlights:
  • Natural sounding wide-band audio for all applications improves communication.
  • Leverages existing IT infrastructure to reduce telecommunications costs.
  • Web-based management access from both the VoIP LAN and
  • AV LAN speed up the work of AV and/or VoIP specialists.
  • Backwards-compatible API saves time when integrating with existing control systems.
  • The card can support two independent rooms.
  • Each line supports two callers for conferencing without an external conference bridge.
  • Composer programming software supports DTMF decoding and call status logic.

Audio:
  • Codecs: G.722, G.711 (A-law and μ-Law), G.729 and G.723.1.
  • DTMF tone generation (RFC 2833, SIP Info and in-band).
  • Low-delay audio packet transmission.
  • Adaptive jitter buffers.
  • Independent level adjustments for call progress, DTMF and ring tones.
  • Country-specific tone compatibility
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