- 2.33 Gbps wireless throughput and 2 Gigabit wireline ports
- Dual-band 4×4:4 MU-MIMO technology
- Self power adaptation upon auto detection of PoE or PoE+
- Supports 200+ concurrent Wi-Fi client devices
- Up to 175-meter coverage range
- Advanced QoS to ensure real-time performance of low-latency applications
- Anti-hacking secure boot and critical data/control lockdown via digital signatures, unique security certificate/ random default password per device
- Embedded controller can manage up to 50 local GWN series APs; GWN.Cloud offers unlimited AP management
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- 2.33Gbps wireless throughput and 2x Gigabit wireline ports
- Dual-band 4x4:4 MIMO technology
- Self power adaptation upon auto detection of PoE/PoE+
- Support 200+ concurrent Wi-Fi client devices
- Up to 300-meter coverage range
- Advanced QoS to ensure real-time performance of low-latency applications
- Anti-hacking secure boot and critical data/control lockdown
- Flexibilty of detachable/ changeable antenna for different application scenarios
- Embedded controller can manage up to 50 local GWN series APs; GWN. Cloud offers unlimited AP management
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- 1.77Gbps aggregate wireless throughput and 2x Gigabit wireline ports
- Dual-band 2×2:2 MU-MIMO & OFMDA Wi-Fi 6 technology
- Self-power adaptation upon auto detection of PoE or PoE+
- Support 256 concurrent Wi-Fi client devices
- Up to 175-meter coverage range 32 SSID total, 16 Per Radio (2.4Ghz and 5Ghz)
- Advanced QoS to ensure real-time performance of low-latency applications
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- 1 SIP account, 2 line keys, 3-way conferencing, 3 XML programmable context-sensitive soft keys
- Dual-switched 10/100 mbps ports
- EHS support for Plantronics headsets
- Up to 500 contacts
- Call history up to 200 records
- Integrated PoE (on GXP1615)
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- 2 SIP accounts, 2 line keys, 3-way conferencing, 3 XML programmable context-sensitive soft keys
- Dual-switched 10/100 mbps ports, integrated PoE on GXP1625
- HD audio on speakerphone and handset
- EHS support for Plantronics headsets
- Up to 500 contacts, call history up to 200 records
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- 2 SIP accounts, 2 line keys, 3-way conferencing, 3 XML programmable context-sensitive soft keys
- HD audio on speakerphone and handset
- Dual-switched Gigabit ports, integrated PoE
- 8 dual-colored BLF/speed dial keys
- EHS support for Plantronics headsets
- Up to 500 contacts, call history up to 200 records
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- 3 SIP accounts, 3 line keys, 4-way conferencing, 3 XML programmable context-sensitive soft keys
- HD audio on speakerphone and handset
- Dual-switched Gigabit ports, integrated PoE
- 8 dual-colored BLF/speed dial keys
- EHS support for Plantronics headsets
- Up to 500 contacts, call history up to 200 records
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- 6 lines, 6 dual-color line keys (with 3 SIP accounts), 4 XML programmable contextsensitive soft keys
- 24 digitally programmable & customizable BLF/fastdial keys
- HD wideband audio, full-duplex hands-free speakerphone with advanced acoustic echo cancellation
- 5-way audio conferencing for easy conference calls
- Dual 10/100 Mbps ports, built-in PoE
- Built-in USB port for importing and exporting data only
- Supports headset use with a RJ9 headset jack and EHS support for EHS-capable Plantronics headsets
- TLS and SRTP security encryption technology to protect calls & accounts and Kensington Security Slot support
- Large phonebook capacity with up to 2,000 contacts and call history with up to 500 records
- Use with Grandstream’s UCM series IP PBX appliance for Zero- Config provisioning, also supports automated provisioning using TR-069 or AES encrypted XML configuration file
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- 6 lines, 6 dual-color line keys (with 3 SIP accounts), 4 XML programmable context-sensitive soft keys
- 5-way audio conferencing for easy conference calls
- 24 digitally programmable & customizable BLF/fast-dial keys
- Built-in USB port for importing and exporting data only
- HD wideband audio, full-duplex hands-free speakerphone with advanced acoustic echo cancellation
- Built-in PoE to power the devices and give it a network connection
- Supports EHS compatible Plantronics’s headsets
- Automated provisioning using TR-069 or AES encrypted XML configuration file
- Large phonebook capacity with up to 2000 contacts and call history with up to 500 records
- TLS and SRTP security encryption technology to protect calls & accounts and Kensington Security Slot support
- Use with Grandstream’s UCM series IP PBX appliance for Zero-Config provisioning, 1-touch call recording & more
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- 8 lines, 8 dual-color line keys (with 4 SIP accounts), 4 XML programmable context sensitive soft keys
- 32 digitally programmable & customizable BLF/fastdial keys
- HD wideband audio, fullduplex speakerphone with advanced acoustic echo cancellation
- 5-way audio conferencing for easy conference calls
- Dual-switched Gigabit ports (GXP1782) or dual-switched 10/100Mbps ports (GXP1780), integrated PoE
- Built-in USB port for importing and exporting data only
- Supports headset use with a RJ9 headset jack and EHS support for EHS-capable Plantronics headsets
- TLS and SRTP security encryption technology to protect calls & accounts and Kensington Security Slot support
- Large phonebook capacity with up to 2,000 contacts and call history with up to 500 records
- Use with Grandstream’s UCM series IP PBX appliance for Zero- Config provisioning, also supports automated provisioning using TR-069 or AES encrypted XML configuration file
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- Supports 3 lines, 3 SIP accounts and 4-way voice conferencing
- 2.8 inch (320x240) color-screen LCD
- Dual Gigabit ports, integrated PoE
- 8 dual-colored BLF/speed-dial keys
- HD audio on speakerphone and handset
- Integrated Bluetooth
- 4 programmable context-sensitive soft keys
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- 8 lines, 4 SIP accounts, 4 XML programmable context-sensitive soft keys
- Dual switched, auto-sensing Gigabit ports, built-in PoE
- 32 digitally programmable and customizable BLF/speed-dial keys
- Built-in Bluetooth for syncing headsets and mobile devices for contact books, calendars & call transferring
- HD audio on the handset and speakerphone; full duplex speakerphone
- Supports EHS compatible Plantronics headsets
- 4-way audio conferencing for easy conference calls
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- Supports 4 lines, 4 SIP accounts and 5-way voice conferencing
- HD audio on speakerphone and handset, 5 programmable context-sensitive soft keys
- Compatible with GXP2200 LCD extension module (GXP2200 EXT)
- 4.3 inch (480x272) color-screen LCD
- Dual Gigabit ports, integrated PoE
- Integrated Bluetooth for use with Bluetooth headsets and Bluetooth-enabled mobile devices (transferring contact books and calls), USB, EHS for Plantronics headsets
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- Supports 6 lines, 6 SIP accounts and 5-way voice conferencing
- 4.3 inch (480x272) color-screen LCD
- Dual Gigabit ports, integrated PoE
- HD audio on speakerphone and handset, 5 programmable context-sensitive soft keys
- 24 dual-colored BLF/speed dial keys
- Integrated Bluetooth
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- 12 lines, 6 SIP accounts, 5 soft keys and 5-way voice conferencing
- 48 on-screen digitally customizable BLF/speed-dial keys
- 4.3 inch (480x272) color-screen LCD
- Dual Gigabit ports, integrated PoE
- Integrated Bluetooth
- Supports up to four GXP2200EXT Modules for BLF/speed-dial access to up to 160 contacts
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- 128x384 backlit LCD display
- 20 programmable dual-color buttons per module, 2 pages per module (40 contacts total)
- BLF/speed dial
- Daisy-chain up to 4 modules for up to 160 contacts/extensions
- BLA (bridged line appearance)/SCA (shared call appearance), BLF (busy lamp field, standard or eventlist), Call Park/Pick-up, Speed Dial, Presence, Intercom, and conference/ transfer/forward and more
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- 16 lines with up to 16 SIP accounts
- Built-in 1 mega-pixel CMOS tiltable camera for video calling with privacy wheel
- Runs on the Android 7.0 operating system
- Built-in Bluetooth for syncing with mobile devices and connecting Bluetooth headsets
- Dual-switched auto-sensing 10/100/1000Mbps network ports
- Integrated dualband Wi-Fi (2.4GHz & 5GHz)
- Built-in PoE/PoE+ for power and network connections
- Dual-mic HD speakerphone with noise reduction, advanced echo cancellation & excellent doubletalk performance
- 4-core 1.3GHz ARM Cortex A53 processor with 2GB RAM and 8GB eMMC Flash
- 5.0” (1280×720) capacitive 5-point touch screen HD TFT LCD
- Peripherals include HDMI-out, USB, headset jack, EHS
- 6-way audio conferencing & 3-way 720p 30fps HD video conferencing capability
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- 16 lines with up to 16 SIP accounts
- Built-in mega-pixel camera for video calling with privacy shutter
- Runs on Android 7.0 operating system
- Built-in Bluetooth for syncing with mobile devices and connecting Bluetooth headset
- Dual-switched auto-sensing 10/100/1000Mbps network ports
- Integrated dual-band WiFi (2.4GHz & 5GHz)
- Built-in PoE/PoE+ to power the device and give it a network connection
- Speakerphone with HD acoustic chamber, advanced echo cancellation & excellent double-talk performance
- 4-core 1.3GHz ARM Cortex A53 processor with 2GB RAM and 8GB eMMC Flash
- 7’’ (1024×600) capacitive 5-point touch screen TFT LCD
- TLS and SRTP security encryption technology to protect calls and accounts
- 7-way audio conferencing & 3-way 720p 30fps HD video conferencing capability
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- 16 lines with up to 16 SIP accounts
- Built-in 2 megapixel camera for video calling with privacy shutter
- Runs on the Android 7.x operating system
- Built-in Bluetooth for syncing with mobile devices and connecting Bluetooth headsets
- Dual-switched auto-sensing 10/100/1000Mbps network ports
- Integrated dualband Wi-Fi (2.4GHz & 5GHz)
- Built-in PoE/PoE+ for power and network connections
- Dual-mic HD speakerphone with advanced echo cancellation & excellent double-talk performance for any scenario
- 64-bit quad-core processor, 2GB RAM, and 16GB Flash
- 8’’ (1280x800) capacitive 10-point touch screen IPS LCD
- Peripherals include HDMI-in/out, USB, Micro SD, headset jack, EHS (Plantronics headsets)
- 7-way HD audio conferencing & 3-way 1080p 30fps HD video capability
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- 4 or 8 FXO PSTN ports
- 2 10/100 Mbps network ports
- 3-way voice conferencing
- Comprehensive codec support, caller ID, flexible dial plans and security protection
- Advanced security protection with SRTP
- Designed and tested for full interoperability with leading IP-PBXs, soft-switches and SIP-based environments
- Failover SIP server feature in case main SIP server goes down
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- 16/24/32 FXS ports, GXW4248 includes 2 50-pin Telco connectors
- 1 Gigabit network port
- 132x48 backlit graphic display with support for multiples languages
- 4 SIP server profiles per system, independent SIP account per port
- Designed and tested for full interoperability with leading IP-PBXs, soft-switches and SIP-based environments
- Advanced security protection with SRTP/TLS/HTTPS
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• Integrates digital PSTN and ISDN trunks with VoIP networks
• 1/2/4 Port E1/T1/J1 spans
• 30/60/120 concurrent calls
• Supports PRI, SS7 and MFC R2 digital signaling
• Supports all popular voice codecs, including Opus, G.722, G.729, iLBC, GSM-FR and more • Support T.38 Fax for creating Fax-Over-IP
• Dual Gigabit ports, configurable NAT router
• 2 USB ports, SD card slot (extra memory)
• Multi-language voice prompts
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- Connects and constantly monitors two UCM6510 together for high availability
- Smart failover solution that automatically switches to a hot-standby secondary UCM6510 if the primary one fails
- Up to 14 LED indicators showing real-time status of all of the telecom lines, network links, auxiliary devices, etc
- Gratuitous ARP forces SIP endpoints to refresh the MAC address of the new UCM6510 without interruptions
- Fast 10 to 50 second system switching time depending on the number of registered endpoints
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- Supports 1 SIP profile through a single FXS port and a single 10/100Mbps port
- TLS and SRTP security encryption technology to protect calls and accounts
- Automated provisioning options include TR-069 and XML config files
- Supports 3-way voice conferencing
- Failover SIP server automatically switches to secondary server if main server loses connection
- Supports T.38 Fax for creating Fax-over-IP
- Supports a wide range of caller ID formats
- Use with Grandstream’s UCM series of IP PBXs for Zero Configuration provisioning
- Supports advanced telephony features, including call transfer, call forward, call-waiting, do not disturb, message waiting indication, multilanguage prompts, flexible dial plan and more
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- Supports 2 SIP profiles through 2 FXS ports and a single 10/100Mbps port
- TLS and SRTP security encryption technology to protect calls and accounts
- Automated provisioning options include TR-069 and XML config files
- Supports 3-way voice conferencing
- Failover SIP server automatically switches to secondary server if main server loses connection
- Supports T.38 Fax for creating Fax-over-IP
- Supports a wide range of caller ID formats
- Use with Grandstream’s UCM series of IP PBXs for Zero Configuration provisioning
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- Supports 2 SIP profiles through 2 FXS ports and dual Gigabit ports
- Includes a built-in NAT router which can handle routing speeds up to 100MBps
- TLS and SRTP security encryption technology to protect calls and accounts
- Automated provisioning options include TR-069 and XML config files
- Supports 3-way voice conferencing
- Failover SIP server automatically switches to secondary server if main server loses connection
- Supports T.38 Fax for creating Fax-over-IP
- Supports a wide range of caller ID formats
- Use with Grandstream’s UCM series of IP PBXs for Zero Configuration provisioning
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- Supports 2 SIP profiles through 1 FXS port and 1 FXO port
- Dual 100Mbps LAN and WAN ports
- Lifeline support (FXS port will be hardrelayed to FXO port)in case of power outage
- 3-way voice conferencing per port
- Automated & secure provisioning options using TR069
- Supports T.38 Fax for reliable Faxover-IP
- Failover SIP server automatically switches to secondary server if main server loses connection
- Strong AES encryption with security certificate per unit
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- Supports 2 SIP profiles through 4 FXS ports and dual Gigabit ports
- Includes a built-in NAT router which can handle routing speeds up to 100MBps
- TLS and SRTP security encryption technology to protect calls and accounts
- Automated provisioning options include TR-069 and XML config files
- Supports 3-way voice conferencing
- Failover SIP server automatically switches to secondary server if main server loses connection
- Supports T.38 Fax for creating Fax-over-IP
- Supports a wide range of caller ID formats
- Use with Grandstream’s UCM series of IP PBXs for Zero Configuration provisioning
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- Supports 2 SIP profiles and 8 FXS ports
- Strong AES encryption with security certificate per unit
- Automated & secure provisioning options using TR069
- 3-way voice conferencing per port
- Exceptional voice quality with wide-band HD codec
- Supports T.38 Fax for reliable Fax-over-IP
- Supports dual Gigabit network ports
- High performance NAT router
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- 50/100/150 Participants.
- .• 49 Video Feeds at a time. • 720P/1080p Full HD Video & HD Audio • Unlimited 1-1 Call • Unlimited Number of Meetings • Access from Any Device • (Windows, Mac, Android ™ or iOS) • Minimum 70Min at Free Plan and Up to 6 Hours Per Group Meeting at paid Version • Screen Sharing, Chats, • Reports/Analytics, Recording • 1-Click Meetings, No Client Downloads • YouTube and Facebook Live Integration • 512MB/2GB/5GB Cloud Recording/Storage • VoIP Dial-In, Toll Dial-In from Land Line or Cell Phone • Group Accounts Administration & Billing • Unlimited Webinars
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- Support for up to 300 participants and 120 video feeds per session; up to 10 simultaneous sessions
- Video and audio recording with 500GB local storage
- 1080p HD at 30fps via H.264/VP8 for real-time video and screen sharing
- Advanced meeting controls, • Flexible scheduling, customizable registration options, follow-up email options, meeting reports and more
- Access from PC/Mac, mobile devices, video conferencing systems, video phones, PSTN trunk, or SIP PBX
- HTTPS and WSS/DTLSSRTP encryption for WebRTC, TLS/SRTP encryption for SIP
- Advanced anti-jitter algorithm to sustain smooth audio & video against up to 30% packet loss
- Live broadcast using Facebook/YouTube Live features
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- UCM6202 and UCM6204 support up to 500 users and 50/75 concurrent calls, UCM6208 supports up to 800 users and 100 concurrent calls
- Auto Discovery and Zero Configuration of Grandstream SIP endpoints
- Integrated 2/4/8 PSTN trunk FXO ports, 2 analog telephone FXS ports with lifeline capability and up to 50 SIP trunk accounts
- Gigabit network ports with Integrates PoE, USB, SD card
- Supports up to a 5-level IVR (Interactive Voice Response)
- Built-in call recordings server; recordings accessible via web user interface
- Built-in Call Detail Records (CDR) for tracking phone usage by line, date, etc.
- Supports multi-language auto-attendant and call queue to efficiently handle incoming calls
- Strongest possible security protection using SRTP, TLS and HTTPS encryption
- Supports any SIP video endpoint that uses the H.264, H.263 or H.263+ codecs
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Supports from 500 to 3000 users Supports from 75 to 450 concurrent calls Zero configuration provisioning of Grandstream SIP endpoints Security protection using SRTP, TLS and HTTPS encryption technology to protect calls and accounts Three Gigabit auto-sensing RJ45 network ports with integrated PoE+ and support NAT router API available for third-party integrations, including CRM and PMS platforms Built-in video conferencing platform with support for Grandstream Wave desktop app Support for wideband HD codecs OPUS & G.722; jitter resilience up to 50% audio packet loss Built-in call center suite and call queue for efficient call volume management Integrated Call Detail Records and call recording server 5-level IVR and multi-language auto-attendant to efficiently handle incoming calls Compatible with GDMS for cloud setup, management and monitoring Supports any SIP video endpoint that using the H.264, H.263 and H.263+ and VP8 codecs -
- UCM6301support up to 500 users and 75 concurrent calls
- Auto Discovery and Zero Configuration of Grandstream SIP endpoints
- Integrated 1 PSTN trunk FXO ports, 1 analog telephone FXS ports with lifeline capability and up to 50 SIP trunk accounts
- Three Gigabit auto-sensing RJ45 network ports with integrated PoE+ and support NAT router, USB, SD card etc.
- Supports up to a 5-level IVR (Interactive Voice Response)
- Built-in call recordings server; recordings accessible via web user interface
- Built-in Call Detail Records (CDR) for tracking phone usage by line, date, etc.
- Supports multi-language auto-attendant and call queue to efficiently handle incoming calls
- Strongest possible security protection using SRTP, TLS and HTTPS encryption
- Supports any SIP video endpoint that uses the H.264, H.263 or H.263+, H.265 codecs
- Compatible with GDMS for cloud setup, management and monitoring
- Automated NAT firewall traversal service facilitates secure remote connections
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- Auto Discovery and Zero Configuration of Grandstream SIP endpoints
- Integrated 2 PSTN trunk FXO ports, 2 analog telephone FXS ports with lifeline capability and up to 50 SIP trunk accounts
- Three Gigabit auto-sensing RJ45 network ports with integrated PoE+ and support NAT router, USB, SD card etc.
- Supports up to a 5-level IVR (Interactive Voice Response)
- Built-in call recordings server; recordings accessible via web user interface
- Built-in Call Detail Records (CDR) for tracking phone usage by line, date, etc.
- Supports multi-language auto-attendant and call queue to efficiently handle incoming calls
- Strongest possible security protection using SRTP, TLS and HTTPS encryption
- Supports any SIP video endpoint that uses the H.264, H.263 or H.263+, H.265 codecs
- Compatible with GDMS for cloud setup, management and monitoring
- Automated NAT firewall traversal service facilitates secure remote connections
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- UCM6304 supports up to 2000 users and 300 concurrent calls
- Auto Discovery and Zero Configuration of Grandstream SIP endpoints
- Integrated 4 PSTN trunk FXO ports, 4 analog telephone FXS ports with lifeline capability and up to 50 SIP trunk accounts
- Three Gigabit auto-sensing RJ45 network ports with integrated PoE+ and support NAT router, USB, SD card etc.
- Supports up to a 5-level IVR (Interactive Voice Response)
- Built-in call recordings server; recordings accessible via web user interface
- Built-in Call Detail Records (CDR) for tracking phone usage by line, date, etc.
- Supports multi-language auto-attendant and call queue to efficiently handle incoming calls
- Strongest possible security protection using SRTP, TLS and HTTPS encryption
- Supports any SIP video endpoint that uses the H.264, H.263 or H.263+, H.265 codecs
- Compatible with GDMS for cloud setup, management and monitoring
- Automated NAT firewall traversal service facilitates secure remote connections
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- UCM6308 supports up to 3000 users and 450 concurrent calls
- Auto Discovery and Zero Configuration of Grandstream SIP endpoints
- Integrated 8 PSTN trunk FXO ports, 8 analog telephone FXS ports with lifeline capability and up to 50 SIP trunk accounts
- Three Gigabit auto-sensing RJ45 network ports with integrated PoE+ and support NAT router, USB, SD card etc.
- Supports up to a 5-level IVR (Interactive Voice Response)
- Built-in call recordings server; recordings accessible via web user interface
- Built-in Call Detail Records (CDR) for tracking phone usage by line, date, etc.
- Supports multi-language auto-attendant and call queue to efficiently handle incoming calls
- Strongest possible security protection using SRTP, TLS and HTTPS encryption
- Supports any SIP video endpoint that uses the H.264, H.263 or H.263+, H.265 codecs
- Compatible with GDMS for cloud setup, management and monitoring
- Automated NAT firewall traversal service facilitates secure remote connections
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- Supports up to 2000 SIP endpoint registrations, up to 200 concurrent calls and up to 64 conference attendees
- 1GHz quad-core Cortex A9 processor
- 1GB DDR3 Ram, 32GB Flash
- 1 Integrated T1/E1/J1 interface, 2PSTN trunk FXO ports, 2 analog telephone/Fax FXS ports with lifeline capability
- Gigabit network ports with Integrates PoE, USB, SD card, integrated NAT router
- Comprehensive security protection using SRTP, TLS and HTTPS with hardware encryption accelerator
- Quickly setup and provision Grandstream endpoints using the Auto-Discovery and Zero Config feature within the product’s web user interface
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- Dual-band Wi-Fi with efficient antenna design and advanced roaming support
- 2 SIP accounts, 2 lines
- HD voice & dual MIC design with AEC and Noise Shield Technology
- Rechargeable 1500mAh battery, 6-hour talk time, 120-hour standby
- Configurable button for push-to-talk
- Micro USB port and 3.5mm headset jack
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- Dual-band WiFi with efficient antenna design and advanced roaming support
- Bluetooth for syncing headsets and mobile devices (contacts and call transferring)
- HD voice & dual MIC design with AEC and Noise Shield Technology
- Rechargeable 1500mAh battery, 7.5 hour talk time, 150-hour standby
- Accelerometer and configurable button for push-to-talk, panic and other related functions
- Micro USB port and 3.5mm headset jack
- 2 SIP accounts, 2 lines
- Supports custom Android apps that fit the phone’s screen/keys
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- Excellent data capture performance on 1D and 2D barcodes, Support NFC for data communication, Supports fingerprint recognition
- 7-inch HD screen, support multi-touch capacitive, cover with 2.5D Corning 3rd generation glass
- Dual-band WiFi with efficient antenna design and fast roaming support
- Rechargeable 4800mAh battery, 12-hour talk time, 200-hour standby; USB Type-C port, supports fast charging
- IP67 dustproof and waterproof, Drop-safe from 1.2-meter height.
- Bluetooth for pairing headsets and mobile devices
- HD voice design with AEC and Noise Shield Technology
- 6 SIP accounts, 6 lines
- Ergonomic design, 12.6mm slim
- Android 9.0 OS, supports custom Android apps
