UCM6302 – Grandstream IP PBX System Model UCM6302 for Voice & Video Collaboration

The Grandstream UCM6302 is an IP PBX system designed to empower businesses with powerful and scalable unified communication and collaboration solutions. As part of the UCM6300 series, it provides a centralized platform for all business communications, including voice, video calling, video conferencing, video surveillance, web meetings, data, analytics, mobility, facility access, intercoms, and more. The UCM6302 supports up to 1000 users and features a built-in web meetings and video conferencing solution, allowing employees to connect from desktops, mobile devices (via the Wave app), GVC series devices, and IP phones. It can be paired with the UCM6300 ecosystem to offer a hybrid platform that combines the control of an on-premise IP PBX with the remote access capabilities of a cloud solution. The UCM6302 also offers advanced security features, easy provisioning, and cloud management through GDMS.

Key Features of UCM6302

• Supports up to 1,000 users and up to 200 concurrent calls (max 120 SRTP concurrent calls)
• Analog Telephone FXS Ports: 2 RJ11 Ports with lifeline capability
• PSTN Line FXO Ports: 2 RJ11 Ports with lifeline capability
• Network Interfaces: Three auto-sensing Gigabit Ethernet ports with integrated PoE+ and NAT router support
• Peripheral Ports: 1 USB 2.0 port, 1 USB 3.0 port, 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 80 audio conference participants and up to 20 video conference participants (in 3 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