• Supports up to 1500 users and up to 200 concurrent calls
• Zero configuration provisioning of Grandstream SIP endpoints
• Built-in Instant Messaging, Audio Conferencing & Web Meetings platform supporting access from computers, mobile devices, and SIP endpoints
• Automated NAT firewall traversal service facilitates secure remote connections
• API available for third-party integrations, including CRM and PMS platforms
• Three Gigabit auto-sensing RJ45 network ports with integrated PoE+ and support NAT router
• Advanced security protection with secure boot, unique certificate, and random default password to protect calls and accounts
• Wave for Android, iOS, Chrome, and Firefox browsers allows communication with all UCM6300 users & solutions
• Supports Full-Band Opus voice codec and H.264/H.263/ H.263+/H.265/VP8 video codec, jitter resilience up to 50% packet loss
• Compatible with GDMS for cloud setup, management, and monitoring
• Based on Asterisk* version 16 open source telephony operating system