Ethernet interface and 2 ISDN BRI interfaces and up to 4 voice channels.

Parlay VoXip Standard is the cost-efficient choice for ITSP?s. It reliably connects any kind of ISDN-equipment to the SIP service, allowing customers to keep single ISDN lines.

• The Parlay VoXip ensures DDI to PBX extensions by individual SIP accounts as well as in accountless mode.

• Its well-proven ISDN implementation guarantees satisfied customers.

• It enables migration from ISDN to low cost VoIP without replacing or upgrading the existing ISDN voice communication platform.

Key Features

· Makes any ISDN PBX VoIP capable ? no modification of the PBX

· Supports call routing to / from SIP based VoIP Service Providers

· VPN: Transparent tunnelling of ISDN and QSIG via IP between 8 Endpoints

· Advanced call routing schemes via ISDN, SIP, VPN

· Seamless re-routing of calls via ISDN if no VoIP connectivity

· Well-proven ISDN stack, handles many national ISDN dialects

• ISDN protocols
  Euro-ISDN protocol according to ETS 300 102.
  Point-Point and Point-Multipoint, including line hunting.
  En-bloc and Overlap sending / receiving.
  QSIG tunneling.
  Supports national dialects: VN2 ? VN6 (France), 1TR6 (Germany),
  BAKOM (Switzerland), INS64 (Japan).
  
  
• Voice Codecs and features
  G.711 (64 kbit/s PCM), A-/?-law
  G.726 (32 kbit/s ADPCM), A-/?-law
  Silence Suppression
  Voice Activity Detection
  Comfort Noise Generation.
  Echo cancellation
  Number of simultaneous IP calls:
  4 Calls (VoXip 104)
  8 Calls (VoXip 108)
  16 Calls (VoXip 116)
  Inbound and outbound DTMF support
  
  
• SIP Features
  Supports 5 Proxies simultaneously
  100 accounts (DDI) at the Proxy
  SIP 2.0 (RFC 3261, RFC 2543)
  Outbound Proxy and STUN support
  Re-routing via ISDN in case of poor IP-link
  QoS, IP Type of Service (DSCP marking)
  ISDN tunneling over IP
  Proprietary ISDN and QSIG tunneling over IP between 8 Parlay VoXip endpoints. RTP Multiplexing to reduce IP overhead
  
  
• Voice Switching Features
  Local switching, allowing internal calls between ISDN interfaces of the Parlay VoXip.
  Local call progress tone generation.
  Local Advice of Charge (AOC) generation, also at SIP calls.
  Call Transfer, Call Forwarding.
  Calling Number modification (e.g. present corporate numbers).
  Called Number modification (e.g. add Carrier Selection Code).
  Voice Routing Table specifying handling of calls:
  Least Cost Routing (e.g. corporate and international calls via IP)
  Toll bypass (e.g. via IP to branch and drop out to ISDN there)
  
  
• Programming
  Via built-in web server, providing a Graphical User Interface.
  Via Telnet session, menu-driven user interface.
  Via RS232 using terminal program, menu-driven user interface

What is SIP?

SIP - the Internet standard for person-to-person communication that is expected to become the next large step of Internet usage after email and surfing, especially when broadband is widely deployed. VoIPtalk is proud to be a reseller of the Mediatrix SIP gateway range.

Click here for more information on SIP.

VoXip Data Sheets

Parlay_VoXip_Data_Sheet (PDF)

Parlay VoXip SIP and VPN Data (PDF)

VoXip Range Datasheet (PDF)

More Information

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