UCM6308 – Grandstream IP PBX System Model UCM6308 for Voice & Video Collaboration

The Grandstream UCM6308 is the top-tier model in the UCM6300 series, a powerful and scalable IP PBX platform designed for large businesses requiring comprehensive unified communication and collaboration solutions. The UCM6308 unifies all business communications onto a single centralized network, encompassing voice, video calling, video conferencing, video surveillance, web meetings, data, analytics, mobility, facility access, and intercoms. This model supports up to 3000 users and includes a built-in web meetings and video conferencing solution, allowing employees to connect flexibly from desktops, mobile devices (via the Wave app), GVC series devices, and IP phones. It can be paired with the UCM6300 Ecosystem to create a hybrid platform that combines the control of an on-premise IP PBX with the secure remote access of a cloud solution (via UCM RemoteConnect). The UCM6308 also offers cloud setup and management through GDMS, along with an API for extensive third-party integrations.

Key Features of UCM6308

• Supports up to 3,000 users and up to 450 concurrent calls (max 300 SRTP concurrent calls)
• Analog Telephone FXS Ports: 8 RJ11 Ports with lifeline capability
• PSTN Line FXO Ports: 8 RJ11 Ports with lifeline capability
• Network Interfaces: Three auto-sensing Gigabit Ethernet ports with integrated PoE+ and NAT router support
• Peripheral Ports: 2 USB 3.0 ports and 1 SD card interface
• LCD Display: 320x240 color LCD with touch screen for Shortcut Keys and Scroll Bar
• Zero-configuration provisioning of Grandstream SIP endpoints
• Built-in conferencing & meetings platform: Supports up to 150 audio conference participants and up to 30 video conference participants (in 6 rooms)
• Wave app for Android, iOS, Chrome, and Firefox browsers enables seamless remote communication and collaboration
• API available for third-party integrations, including CRM and PMS platforms
• Advanced security protection with secure boot, unique certificate, and random default password
• Automated NAT firewall traversal service for secure remote connections
• Supports Full-Band Opus voice codec and H.264/H.263/H.263+/H.265/VP8 video codec, with 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
• Voice-over-Packet Capabilities: LEC, Dynamic Jitter Buffer, Modem detection & auto-switch to G.711, NetEQ, FEC 2.0
• DTMF Methods: In-band audio, RFC2833, and SIP INFO
• Media Encryption: SRTP, TLS, HTTPS, SSH, 802.1X, ZRTP
• Supports both wall mount and desktop installation