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UCM6308
Rated 0 out of 5Features
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(✓) Supports 3000 users and 450 concurrent calls and Max 300 concurrent SRTP calls.
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(✓) Zero configuration provisioning of Grandstream SIP endpoints.
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(✓) 10 Video Conference rooms and up to 80 parties with 1080p, assuming 4 video feeds + 1 screen sharing (H.264 & Opus) Voice Conference: Up to 300 parties.
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(✓) Built-in conferencing & meetings platform; supports desktop, and SIP endpoints.
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(✓) Wave for Android, iOS, Chrome and Firefox browsers allows communication with all UCM6300 users & solutions.
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(✓) Advanced security protection with secure boot, unique certificate and random default password to protect calls and accounts.
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(✓) Three Gigabit auto-sensing RJ45 network ports with integrated PoE+ and support NAT router.
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(✓) Automated NAT firewall traversal service facilitates secure remote connections.
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(✓) 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.
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(✓) Compatible with GDMS for cloud setup, management and monitoring.
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(✓) Based on Asterisk* version 16 open source telephony operating system.
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UCM6308A
Rated 0 out of 5Features
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(✓) Supports 1500 users and 200 concurrent calls and Max 150 concurrent SRTP calls.
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(✓) Zero configuration provisioning of Grandstream SIP endpoints.
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(✓) 9 meeting rooms and up to 150 parties.
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(✓) Built-in Instant Messaging (IM), Audio Conferencing & Web Meetings platform that supports access from computers, mobile devices, and SIP endpoints.
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(✓) Advanced security protection with secure boot, unique certificate and random default password to protect calls and accounts.
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(✓) Three Gigabit auto-sensing RJ45 network ports with integrated PoE+ and support NAT router.
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(✓) Automated NAT firewall traversal service facilitates secure remote connections.
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(✓) Enhanced reliability with support for Hot Standby High-Availability and local dual deployment.
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(✓) Supports Full-Band Opus voice codec, jitter resilience up to 50% packet loss.
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(✓) Compatible with GDMS for cloud setup, management and monitoring.
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(✓) Based on Asterisk* version 16 open source telephony operating system.
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UCM6304
Rated 0 out of 5Features
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(✓) Supports 2000 users and 300 concurrent calls and Max 200 concurrent SRTP calls.
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(✓) Zero configuration provisioning of Grandstream SIP endpoints.
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(✓) 8 Video Conference rooms and up to 60 parties with 1080p, assuming 4 video feeds + 1 screen sharing (H.264 & Opus) Voice Conference: Up to 200 parties .
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(✓) Built-in conferencing & meetings platform; supports desktop, and SIP endpoints.
-
(✓) Wave for Android, iOS, Chrome and Firefox browsers allows communication with all UCM6300 users & solutions.
-
(✓) Advanced security protection with secure boot, unique certificate and random default password to protect calls and accounts.
-
(✓) Three Gigabit auto-sensing RJ45 network ports with integrated PoE+ and support NAT router.
-
(✓) Automated NAT firewall traversal service facilitates secure remote connections.
-
(✓) 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.
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(✓) Compatible with GDMS for cloud setup, management and monitoring.
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(✓) Based on Asterisk* version 16 open source telephony operating system.
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UCM6304A
Rated 0 out of 5Features
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(✓) Supports 1000 users and 150 concurrent calls and Max 120 concurrent SRTP calls.
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(✓) Zero configuration provisioning of Grandstream SIP endpoints.
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(✓) 7 meeting rooms and up to 120 parties.
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(✓) Built-in Instant Messaging (IM), Audio Conferencing & Web Meetings platform that supports access from computers, mobile devices, and SIP endpoints.
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(✓) Advanced security protection with secure boot, unique certificate and random default password to protect calls and accounts.
-
(✓) Three Gigabit auto-sensing RJ45 network ports with integrated PoE+ and support NAT router.
-
(✓) Automated NAT firewall traversal service facilitates secure remote connections.
-
(✓) Enhanced reliability with support for Hot Standby High-Availability and local dual deployment.
-
(✓) Supports Full-Band Opus voice 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.
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UCM6302
Rated 0 out of 5Features
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(✓) Supports 1000 users and 150 concurrent calls and Max 100 concurrent SRTP calls.
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(✓) Zero configuration provisioning of Grandstream SIP endpoints.
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(✓) 6 Video Conference rooms and up to 30 parties with 1080p, assuming 4 video feeds + 1 screen sharing (H.264 & Opus) Voice Conference: Up to 150 parties .
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(✓) Built-in conferencing & meetings platform; supports desktop, and SIP endpoints.
-
(✓) Wave for Android, iOS, Chrome and Firefox browsers allows communication with all UCM6300 users & solutions.
-
(✓) Advanced security protection with secure boot, unique certificate and random default password to protect calls and accounts.
-
(✓) Three Gigabit auto-sensing RJ45 network ports with integrated PoE+ and support NAT router.
-
(✓) Automated NAT firewall traversal service facilitates secure remote connections.
-
(✓) 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.
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(✓) Compatible with GDMS for cloud setup, management and monitoring.
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(✓) Based on Asterisk* version 16 open source telephony operating system.
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UCM6302A
Rated 0 out of 5Features
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(✓) Supports 500 users and 75 concurrent calls and Max 75 concurrent SRTP calls.
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(✓) Zero configuration provisioning of Grandstream SIP endpoints.
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(✓) 5 meeting rooms and up to 75 parties.
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(✓) Built-in Instant Messaging (IM), Audio Conferencing & Web Meetings platform that supports access from computers, mobile devices, and SIP endpoints.
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(✓) Advanced security protection with secure boot, unique certificate and random default password to protect calls and accounts.
-
(✓) Three Gigabit auto-sensing RJ45 network ports with integrated PoE+ and support NAT router.
-
(✓) Automated NAT firewall traversal service facilitates secure remote connections.
-
(✓) Enhanced reliability with support for Hot Standby High-Availability and local dual deployment.
-
(✓) Supports Full-Band Opus voice codec, jitter resilience up to 50% packet loss.
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(✓) Compatible with GDMS for cloud setup, management and monitoring.
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(✓) Based on Asterisk* version 16 open source telephony operating system.
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UCM6208
Rated 0 out of 5Features
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(✓) UCM6208 supports up to 800 users and 100 concurrent calls
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(✓) Auto Discovery and Zero Configuration of Grandstream SIP endpoints
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(✓) Integrated 8 PSTN trunk FXO ports, 2 analog telephone FXS ports with lifeline capability and up to 200 SIP trunk accounts
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(✓) Gigabit network ports with Integrates PoE, USB, SD card
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(✓) Supports up to a 5-level IVR (Interactive Voice Response)
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(✓) Built-in call recordings server; recordings accessible via web user interface
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(✓) Built-in Call Detail Records (CDR) for tracking phone usage by line, date, etc.
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(✓) Supports multi-language auto-attendant and call queue to efficiently handle incoming calls
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(✓) Strongest possible security protection using SRTP, TLS and HTTPS encryption
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(✓) Supports any SIP video endpoint that uses the H.264, H.263 or H.263+ codecs
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UCM6204
Rated 0 out of 5Features
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(✓) UCM6204 support up to 500 users and 75 concurrent calls
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(✓) Auto Discovery and Zero Configuration of Grandstream SIP endpoints
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(✓) Integrated 4 PSTN trunk FXO ports, 2 analog telephone FXS ports with lifeline capability and up to 200 SIP trunk accounts
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(✓) Gigabit network ports with Integrates PoE, USB, SD card
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(✓) Supports up to a 5-level IVR (Interactive Voice Response)
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(✓) Built-in call recordings server; recordings accessible via web user interface
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(✓) Built-in Call Detail Records (CDR) for tracking phone usage by line, date, etc.
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(✓) Supports multi-language auto-attendant and call queue to efficiently handle incoming calls
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(✓) Strongest possible security protection using SRTP, TLS and HTTPS encryption
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(✓) Supports any SIP video endpoint that uses the H.264, H.263 or H.263+ codecs
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UCM6202
Rated 0 out of 5Features
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(✓) UCM6202 support up to 500 users and 50 concurrent calls
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(✓) Auto Discovery and Zero Configuration of Grandstream SIP endpoints
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(✓) Integrated 2 PSTN trunk FXO ports, 2 analog telephone FXS ports with lifeline capability and up to 200 SIP trunk accounts
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(✓) Gigabit network ports with Integrates PoE, USB, SD card
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(✓) Supports up to a 5-level IVR (Interactive Voice Response)
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(✓) Built-in call recordings server; recordings accessible via web user interface
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(✓) Built-in Call Detail Records (CDR) for tracking phone usage by line, date, etc.
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(✓) Supports multi-language auto-attendant and call queue to efficiently handle incoming calls
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(✓) Strongest possible security protection using SRTP, TLS and HTTPS encryption
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(✓) Supports any SIP video endpoint that uses the H.264, H.263 or H.263+ codecs
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GXV3380
Rated 0 out of 5Features
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(✓) 16 lines, with up to 16 SIP accounts.
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(✓) Built-in 2 mega-pixel camera for video calling with privacy shutter.
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(✓) Runs on the Android 7.0 operating system.
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(✓) Built-in Bluetooth for syncing headsets and mobile devices for contact books, calendars & call transferring.
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(✓) 8.0 inch (1280x800) capacitive touch screen LCD display.
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(✓) Dual-switched auto-sensing 10/100/1000 Gigabit ports.
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(✓) Integrated dual-band WiFi (2.4GHz and 5.0 GHz) and WMM.
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(✓) 64-bit quad-core processor, 2GB RAM, and 16GB Flash.
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(✓) Mini-HDMI outputs for connection to a TV or computer monitor.
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(✓) HD Audio and Video.
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(✓) Dual-mic HD speakerphone with noise reduction, advanced echo cancellation & excellent doubletalk performance.
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(✓) 7-way audio conferencing and 3-way 1080P HD video conferencing.
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(✓) TLS and SRTP security encryption technology to protect calls and accounts.
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(✓) USB port.
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(✓) RJ9 headset jack, allowing EHS with Plantronics headsets.
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(✓) 2 function touch keys (volume +/-) and 3 dedicated Android touch keys (home, menu, back).
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GXV3370
Rated 0 out of 5Features of Grandstream GXV3370
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(✓) 16 lines, with up to 16 SIP accounts.
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(✓) Built-in mega-pixel camera for video calling with privacy shutter.
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(✓) Runs on Android 7.0 operating system.
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(✓) Built-in Bluetooth for syncing headsets and mobile devices for contact books, calendars & call transferring.
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(✓) 7.0 inch (1024x600) capacitive touch screen LCD display.
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(✓) Dual-switched auto-sensing 10/100/1000 Gigabit ports.
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(✓) Integrated dual-band Wi-Fi (2.4GHz and 5.0 GHz).
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(✓) 4-core 1.3GHz ARM Cortex A53 processor with 2GB RAM and 8GB eMMC Flash.
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(✓) Mini-HDMI outputs for connection to a TV or computer monitor.
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(✓) HD Audio and Video.
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(✓) HD speakerphone with noise reduction, advanced echo cancellation & excellent doubletalk performance.
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(✓) 7-way audio conferencing and 3-way 720P HD video conferencing.
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(✓) TLS and SRTP security encryption technology to protect calls and accounts.
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(✓) USB port.
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(✓) RJ9 headset jack, allowing EHS with Plantronics headsets.
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(✓) 2 function touch keys (volume +/-) and 3 dedicated Android touch keys (home, menu, back).
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GXV3350
Rated 0 out of 5Features
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(✓) 16 lines, with up to 16 SIP accounts.
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(✓) Built-in 1 mega-pixel CMOS tiltable camera for video calling with privacy wheel.
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(✓) Runs on the Android 7.0 operating system.
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(✓) Built-in Bluetooth for syncing with mobile devices and connecting Bluetooth headsets.
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(✓) 5.0 inch (1280x720) capacitive touch screen LCD display.
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(✓) Dual-switched auto-sensing 10/100/1000 Gigabit ports.
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(✓) Integrated dual-band Wi-Fi (2.4GHz and 5.0 GHz).
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(✓) Mini-HDMI outputs for connection to a TV or computer monitor.
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(✓) HD Audio and Video.
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(✓) Dual-mic HD speakerphone with noise reduction, advanced echo cancellation & excellent doubletalk performance.
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(✓) 6-way audio conferencing and 3-way 720P HD video conferencing.
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(✓) TLS and SRTP security encryption technology to protect calls and accounts.
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(✓) USB port.
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(✓) RJ9 headset jack, allowing EHS with Plantronics headsets.
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(✓) 11 function keys (phonebook, backspace, headset, mute, hold, message, transfer, conference, speakerphone, send/redial, volume) and 3 dedicated android keys (home, menu, and back).
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