Grandstream, the award-winning supplier of unified communications, have announced the release of the new UCM6300 Audio Series. This new unified communication and collaboration solution provides businesses with a robust, scalable platform that includes all features from the state-of-the-art UCM6300 Series, with only video capability removed. This allows the UCM6300 Audio Series to be the ideal communications solution for businesses looking for a powerful-yet cost-effective platform and for those who do not need video.

The UCM6300 Audio Series provides a platform that unifies fundamental business communications and collaboration needs onto one network, including voice, instant messaging (IM), voice meetings, audio web meetings, data, analytics, mobility, facility access, intercoms and more. The UCM6300 Audio Series pairs with the UCM6300 Series Ecosystem, including the Wave app and UCM RemoteConnect cloud service, to provide a hybrid communication platform that offers the control and reduced total cost of ownership of an on-premise IP PBX with the remote-access flexibility of a cloud solution. Other features of the UCM6300 Audio Series include:

  • Support for up to 1500 users and up to 200 concurrent calls
  • Zero configuration provisioning of Grandstream SIP endpoints
  • Built-in Instant Messaging (IM), Audio Conferencing & Web Meetings platform that supports access from computers, mobile devices, and SIP endpoints
  • Free Wave App allows easy voice & Instant Messaging (IM) communications using desktops, Web, and Android/ iOS devices
  • Pairs with the UCM RemoteConnect cloud service, which provides an automated NAT firewall traversal service to facilitate secure remote connections
  • API available for third-party application integrations, including CRM and PMS platforms
  • Advanced security protection with secure boot, unique certificate and random default password to protect calls and accounts
  • Three Gigabit auto-sensing network ports with integrated PoE+ and NAT router, up to 8 FXS analog telephone ports and 8 FXO PSTN ports
  • Enhanced reliability with support for Hot Standby High-Availability and local redundant deployment
  • Supports Full-Band Opus voice codec with 48KHz voice sampling rate, jitter resilience against up to 50% packet loss
  • Compatible with GDMS for cloud setup, management, and monitoring
  • Based on Asterisk version 16 open-source telephony operating system

Expand your solution

Want to expand your communications solution further? Check out the full range of Grandstream products below! If you have any specific requirements, please contact our technical sales team on +44 330 088 0195.

Check out the original story from Grandstream HERE