GXHP610 is compact IP phone that provide an HD speaker on the handset, 2 SIP accounts/lines, 10 speed dial keys and 3 programmable keys. This simple IP phone is also ideal for hospitals, apartments, dormitories, retail and more. This Series includes 2 models with integrated dual-band Wi-Fi support (GHP610W and GHP611W), provides 3-way voice conferencing, supports full-band OPUS voice codec and features an advanced jitter-resilience algorithm that tolerates up to 30% packet loss without impacting voice quality. The GHP Series is supported by Grandstream’s Device Management System (GDMS), which provides a centralized interface to configure, provision, manage and monitor deployments of Grandstream endpoints. Thanks to a compact design that can be easily used on a desktop or wall-mounted

Protocols/Standards SIP: SIP RFC3261, TCP/IP/UDP, RTP/RTCP, RTCP-XR, TLS, SRTP
Network: IPv4, IPv6, HTTP/HTTPS, ARP, ICMP, DNS(A record, SRV, NAPTR), DHCP, SSH, TFTP, NTP, STUN, LLDP, TR-069, 802.1x
Network Interfaces One auto-negotiation 10/100 Mbps ethernet port integrated PoE Class 2 
Camera Tiltable 2 mega-pixel CMOS camera with privacy shutter, 1080P@30fps
Wi-Fi (2.4GHz & 5GHz 802.11 a/b/g/n/ac)
Voice Codecs and Capabilities Support for G.729A/B, G.711µ/a-law, G.726, G.722 (wide-band), G.723, iLBC, and full band OPUS. In- band and out-of-band DTMF (in audio, RFC2833, SIP INFO), VAD, AEC, CNG, PLC, AGC, AJB

GXV3450 High-End Smart IP Video Phone for Android is an advanced desktop video collaboration solution that combines a 16 line IP phone with the functionality of an Android tablet to offer an all-in-one communication solution. It is the ideal desktop device for busy professionals and executives and also offers a powerful yet cost-effective device for any conference room.

Protocols/Standards SIP RFC3261, TCP/IP/UDP, RTP/RTCP, HTTP/HTTPS, ARP, ICMP, DNS (A record, SRV, NAPTR), DHCP, PPPoE, SSH, TFTP, NTP, STUN, LLDP-MED, LDAP, TR-069, 802.1x, TLS, SRTP, IPv6, OpenVPN®
Network Interfaces Dual switched 10/ 100/ 1000 Mbps ports with integrated PoE/PoE+
Camera Tiltable 2 mega-pixel CMOS camera with privacy shutter, 1080P@30fps
Bluetooth, Wi-Fi (2.4GHz & 5GHz)
Auxiliary Ports RJ9 headset jack (allowing EHS with Plantronics headsets), USB 2.0 port, USB 3.0 port
Voice Codecs and Capabilities Wide-band Opus, wide-band G.722, G.711μ/a, G. 729A/B, G.726-32, iLBC, in-band and out-ofband DTMF (In audio, RFC2833, SIP INFO), VAD, CNG, AEC, PLC, AJB, AGC, ANS, Noise Shield 2.0
Video Codecs and Capabilities H.264 BP/MP/HP, video resolution up to 720p, frame rate up to 30 fps, bit rate up to 2Mbps, 3-way video conference (720p@30fps), anti-flickering, and auto exposure

USB RFID Card Reader is available for use with the GDS series of IP Door Systems. This USB RFID Card reader can be installed anywhere and connect via USB to any computer running Grandstream’s free GDS Manager software to allow all RFID Card Reader activity to be tracked, monitored and managed centrally by GDS Manager. The device is fully powered through USB and offers a built-in antenna with card-seeking mode.

Card Supported 125KHz: EM4100 RFID card
Working Frequency 125KHz
Power Supply DC 5V (±4%)
Electric Current Standby Current 20mA; Operation Current Less Than 35 mA
Operating Distance 0-40mm effective distance, the use of ID card and the Reader

RFID Card is available for use with the GDS series and the USB RFID Card Reader. This nonwritable proximity induction card receives radio energy from the RFID Card Reader to power the card and its ability to interact with any GDS series device and the USB RFID Card Reader. This contactless ID card is easy to setup and use, and therefore is ideal for use in a wide range of facility access, attendance and security applications.

Card Type ID Card (TK4100)
Working Frequency 125KHz
Storage Capacity 64 Bit Read-Only
Operating Distance 2 ~ 15 cm
Life Span > 100,000 Times

GRP2603 is an essential 3-line model designed with zerotouch provisioning for mass deployment and easy management. It features a sleek design and a suite of nextgeneration features including: 5-way voice conferencing to maximize productivity, full HD audio on both the speaker and handset to allow users to communicate with the utmost clarity, EHS support for Plantronics, Jabra, and Sennheiser headsets and multi-language support.

Protocols/Standards SIP RFC3261, TCP/IP/UDP, RTP/RTCP, RTCP-XR, HTTP/HTTPS, ARP, ICMP, DNS(A record, SRV, NAPTR), DHCP, PPPoE, SSH, TFTP, NTP, STUN, SIMPLE, LLDP, LDAP, TR069, SNMP, 802.1x, TLS, SRTP, IPv6
Network Interfaces Dual switched auto-negotiation 10/100/1000 Mbps Ethernet ports
Auxiliary Ports RJ9 headset jack (allowing EHS with Plantronics & Jabra &Sennheiser headsets)
Voice Codecs and Capabilities Support for G.729A/B, G.711µ/a-law, G.726, G.722(wide-band), G.723,iLBC, OPUS, in- band and out-of-band DTMF(in audio, RFC2833, SIP INFO), VAD, AEC, CNG, PLC, AGC, AJB

Supervoice SVC-WBT31 Wireless Bluetooth Headset Mono Crystal Clear Sound Quality with directional noise canceling microphone are built with wideband audio to ensure HD natural hearing. The acoustic noise canceling structure ensures crystal-clear conversations even in noisy environment. The One Press mute button is located on the microphone boom tip which makes it much easier to mute during a call. When on the phone or working you can prevent interruptions using the busy light indicator. Luxurious Comfortable & Super Lightweight design for whole day wearing.

Bluetooth Specifications CSR-V5.0, dual mode, downward compatible
Frequency Range 2.4GHz-2.48GHz
Support Protocols A2DP, AVRCP, HFP, AAC, etc.
Transmission distance Up to 30 meters (Varies on base of the environment and Bluetooth device)
Pair with 2 Bluetooth devices simultaneously
Wideband Speaker Frequency 20Hz-20KHz
Audio Monaural/Mono

GRP2603 is an essential 3-line model designed with zerotouch provisioning for mass deployment and easy management. It features a sleek design and a suite of nextgeneration features including: 5-way voice conferencing to maximize productivity, integrated PoE (GRP2603P), full HD audio on both the speaker and handset to allow users to communicate with the utmost clarity, EHS support for Plantronics, Jabra, and Sennheiser headsets and multi-language support.

Protocols/Standards SIP RFC3261, TCP/IP/UDP, RTP/RTCP, RTCP-XR, HTTP/HTTPS, ARP, ICMP, DNS(A record, SRV, NAPTR), DHCP, PPPoE, SSH, TFTP, NTP, STUN, SIMPLE, LLDP, LDAP, TR069, SNMP, 802.1x, TLS, SRTP, IPv6
Network Interfaces Dual switched auto-negotiation 10/100/1000 Mbps Ethernet ports, integrated PoE
Auxiliary Ports RJ9 headset jack (allowing EHS with Plantronics & Jabra &Sennheiser headsets)
Voice Codecs and Capabilities Support for G.729A/B, G.711µ/a-law, G.726, G.722(wide-band), G.723,iLBC, OPUS, in- band and out-of-band DTMF(in audio, RFC2833, SIP INFO), VAD, AEC, CNG, PLC, AGC, AJB

SVC101 Call Center Headset Mono w/o Bottom Cable is the ideal headset device for office and call-center staff. It features a light weight design ensuring exceptional user comfort with 60mm adjustable steel headband, 330° rotatable microphone arm, 180° rotatable joint of speaker and headband. Moreover, it is consisted of a pliable T-bar and a flexible boom. It delivers superb receiver audio quality with excellent noise cancellation technology and remarkable anti-acoustic shock circuit.

Frequency Range 50Hz-3.5KGHz
Drive Style: Dynamic (Moving-coiled)
Wideband Speaker Frequency 300Hz-3.0KHz
Audio Mono

DP730 is a DECT cordless IP phone that allows users to mobilize their VoIP network throughout any business, warehouse, retail store and residential environment. It is supported by Grandstream’s DP750 and DP752 DECT VoIP base stations and delivers a combination of mobility and top-notch telephony performance. Up to five DP730 handsets are supported on each base station while each DP730 supports a range of up to 400 meters outdoors (with DP752) and 50 meters indoors along with 40 hours of talk time and 500-hour standby time.

Protocols/Standards Hearing Aid Compatibility (HAC) compliant
Voice Codecs and Capabilities G.722 codec for HD audio and G.726 codec for narrow band audio (G.711μ/a-law, G.723.1, G.729A/B, iLBC and OPUS are supported via companion DECT base station, AEC, AGC, Ambient noise reduction on handset mic, advanced noise suppression for incoming audio

UCM6300 Audio series allows businesses to build powerful and scalable unified communication and collaboration solutions. This series of IP PBXs provide a platform that unifies fundamental business communications needs, including voice, instant messaging (IM), voice meetings, audio web meetings, data, analytics, mobility, facility access, intercoms and more. The UCM6300 Audio Series supports up to 1500 users and includes a built-in instant messaging (IM), voice/web conferencing platform, and the free Wave App that allows users to communicate and collaborate from desktops, mobile devices, IP phones, and other SIP endpoints. It supports UCM RemoteConnect cloud service for remote users to offer a best-in class hybrid platform that combines the control of an on-premise IP PBX with the remote access and system manageability of a cloud solution.

2 x FXS and 2 x FXO ports
Peripheral Ports: 1*USB 2.0, 1*USB 3.0, 1*SD card interface
Voice and Fax Codecs: Opus, G.711 A-law/U-law, G.722, G722.1 G722.1C, G.723.1 5.3K/6.3K, G.726-32, G.729A/B, iLBC, GSM; T.38
Provisioning Protocol & Plug-and-Play : Mass provisioning using AES encrypted XML configuration file, auto-discovery & auto-provisioning of Grandstream IP endpoints via ZeroConfig (DHCP Option 66 multicast SIP SUBSCRIBE mDNS), eventlist between local and remote trunk
Maximum Call Capacity: Users: 500 Concurrent calls (G.711): 75 Max concurrent SRTP calls (G.711): 75
Maximum Attendees of Conference Bridges: 5 meeting rooms and up to 75 parties

UCM6300 Audio series allows businesses to build powerful and scalable unified communication and collaboration solutions. This series of IP PBXs provide a platform that unifies fundamental business communications needs, including voice, instant messaging (IM), voice meetings, audio web meetings, data, analytics, mobility, facility access, intercoms and more. The UCM6300 Audio Series supports up to 1500 users and includes a built-in instant messaging (IM), voice/web conferencing platform, and the free Wave App that allows users to communicate and collaborate from desktops, mobile devices, IP phones, and other SIP endpoints. It supports UCM RemoteConnect cloud service for remote users to offer a best-in class hybrid platform that combines the control of an on-premise IP PBX with the remote access and system manageability of a cloud solution.

8 x FXS and 8 x FXO ports
Peripheral Ports: 2*USB 3.0, 1*SD card interface
Voice and Fax Codecs: Opus, G.711 A-law/U-law, G.722, G722.1 G722.1C, G.723.1 5.3K/6.3K, G.726-32, G.729A/B, iLBC, GSM; T.38
Provisioning Protocol & Plug-and-Play: Mass provisioning using AES encrypted XML configuration file, auto-discovery & auto-provisioning of Grandstream IP endpoints via ZeroConfig (DHCP Option 66 multicast SIP SUBSCRIBE mDNS), eventlist between local and remote trunk
Maximum Call Capacity: Users: 1500, Concurrent calls (G.711): 200, Max concurrent SRTP calls (G.711): 150
Maximum Attendees of Conference Bridges: 9 meeting rooms and up to 150 parties

UCM6510 creates an easily manageable on premise anchor to your communications network. This enterprise-grade IP PBX comes equipped with a suite of advanced call handling and network data features, all with no licensing and no fees. Its scalability offers deployments that can support up to 2000 users, and it supports E1, T1 and J1. The UCM6510 series allows businesses to unify multiple communication technologies, such as voice, video, surveillance, data tools, and facilities access management into one common platform that can be managed and accessed remotely.

2 x FXS and 2 x FXO ports
Peripheral Ports: USB, SD
Voice and Fax Codecs: G.711 A-law/U-law, G.722, G.723.1 5.3K/6.3K, G.726, G.729A/B, iLBC, GSM, AAL2-G.726-32, ADPCM; T.38
Provisioning Protocol & Plug-and-Play: TFTP/HTTP/HTTPS, auto-discovery & auto-provisioning of Grandstream IP endpoints via ZeroConfig (DHCP
Option 66 multicast SIP SUBSCRIBE mDNS), eventlist between local and remote trunk
Maximum Call Capacity: Up to 2000 registered SIP endpoints, up to 200 concurrent calls
Maximum Attendees of Conference Bridges: Up to 8 bridges, up to 64 simultaneous conference attendees

GRP2602P is an essential 2-line model designed with zero-touch provisioning for mass deployment and easy management. It features a sleek design and a suite of next-generation features, 5-way voice conferencing to maximize productivity, integrated PoE, full HD audio on both the speaker and handset to allow users to communicate with the utmost clarity, EHS support for Plantronics, Jabra, and Sennheiser headsets and multi-language support.

Protocols/Standards SIP RFC3261, TCP/IP/UDP, RTP/RTCP, RTCP-XR, HTTP/HTTPS, ARP, ICMP, DNS(A record, SRV, NAPTR), DHCP, PPPoE, SSH, TFTP, NTP, STUN, SIMPLE, LLDP, LDAP, TR069, SNMP, 802.1x, TLS, SRTP, IPv6
Network Interfaces Dual switched auto-negotiation 10/100 Mbps Ethernet ports, integrated PoE 
Auxiliary Ports RJ9 headset jack (allowing EHS with Plantronics & Jabra &Sennheiser headsets)
Voice Codecs and Capabilities Support for G.729A/B, G.711µ/a-law, G.726, G.722(wide-band), G.723,iLBC, OPUS, in- band and out-of-band DTMF(in audio, RFC2833, SIP INFO), VAD, AEC, CNG, PLC, AGC, AJB

GRP2601P is an essential 2-line model designed with zero-touch provisioning for mass deployment and easy management. It features a sleek design and a suite of next-generation features including 5-way voice conferencing to maximize productivity, integrated PoE, EHS support for Plantronics, Jabra, and Sennheiser headsets and multi-language support.

Protocols/Standards SIP RFC3261, TCP/IP/UDP, RTP/RTCP, RTCP-XR, HTTP/HTTPS, ARP, ICMP, DNS(A record, SRV, NAPTR), DHCP, PPPoE, SSH, TFTP, NTP, STUN, SIMPLE, LLDP, LDAP, TR069, SNMP, 802.1x, TLS, SRTP, IPv6
Network Interfaces Dual switched auto-negotiation 10/100 Mbps Ethernet ports, integrated PoE
Auxiliary Ports RJ9 headset jack (allowing EHS with Plantronics & Jabra &Sennheiser headsets)
Voice Codecs and Capabilities Support for G.729A/B, G.711µ/a-law, G.726, G.722(wide-band), G.723,iLBC, OPUS, in- band and out-of-band DTMF(in audio, RFC2833, SIP INFO), VAD, AEC, CNG, PLC, AGC, AJB

UCM6300 series allows businesses to build powerful and scalable unified communication and collaboration solutions. This series of IP PBXs provide a platform that unifies all business communication on one centralized network, including voice, video calling, video conferencing, video surveillance, web meetings, data, analytics, mobility, facility access, intercoms and more. The UCM6300 series supports up to 3000 users and includes a built-in web meetings and video conferencing solution that allows employees to connect from the desktop, mobile, 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 of a cloud solution.

2 x FXS and 2 x FXO ports
Peripheral Ports: 1*USB 2.0, 1*USB 3.0, 1*SD card interface
Voice and Fax Codecs: Opus, G.711 A-law/U-law, G.722, G722.1 G722.1C, G.723.1 5.3K/6.3K, G.726-32, G.729A/B, iLBC, GSM; T.38
Provisioning Protocol & Plug-and-Play: Mass provisioning using AES encrypted XML configuration file, auto-discovery & auto-provisioning of Grandstream IP endpoints via ZeroConfig
(DHCP Option 66 multicast SIP SUBSCRIBE mDNS), eventlist between local and remote trunk
Maximum Call Capacity: Users: 1000, Concurrent calls (G.711): 150, Max concurrent SRTP calls (G.711): 100
Maximum Attendees of Conference Bridges: 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 (G.711)

GXV3370 is a powerful desktop video phone for enterprise users. It features a 7” touch screen, advanced megapixel camera for HD video conferencing, built-in Wi-Fi and Bluetooth, Gigabit network speeds and innovative telephony functionalities. It also runs on Android 7.0 and has flexible SDK support for custom apps. The GXV3370 is fully interoperable with nearly all major SIP platforms on the market and can be seamlessly integrated with Grandstream’s portfolio including SIP based security cameras, door systems, IP PBXs, and video conferencing systems and services.

Protocols/Standards SIP RFC3261, TCP/IP/UDP, RTP/RTCP, HTTP/HTTPS, ARP, ICMP, DNS (A record, SRV, NAPTR), DHCP, PPPoE, SSH, TFTP, NTP, STUN, SIMPLE, LLDP-MED, LDAP, TR-069, 802.1x, TLS, SRTP, IPv6, OpenVPN®
Network Interfaces Dual switched 10/ 100/ 1000 Mbps ports with integrated PoE/PoE+
Camera Tiltable mega-pixel CMOS camera with privacy shutter, 720p 30fps
Bluetooth, Wi-Fi (2.4GHz & 5GHz)
Auxiliary Ports RJ9 headset jack (allowing EHS with Plantronics headsets), 3.5mm stereo headset with microphone, USB port, SD, HDMI-out (1.4 up to 720p 30fps)
Voice Codecs and Capabilities G.711µ/a, G.722 (wide-band), G.726-32, iLBC, Opus, G.729A/B, in-band and out-of-band DTMF (In audio, RFC2833, SIP INFO), VAD, CNG, AEC, PLC, AJB, AGC, ANS
Video Codecs and Capabilities H.264 BP/MP/HP, video resolution up to 720p, frame rate up to 30 fps, bit rate up to 2Mbps, 3-way video conference (720p 30fps), anti-flickering, auto focus and auto exposure

GXV3350 IP Video Phone for Android combines a 16-line IP video phone with a multi-platform video collaboration solution and the functionality of an Android tablet to offer an all-in-one communications solution. This IP video phone delivers a powerful experience through its’ 5 inch 1280×720 capacitive touch screen, tiltable camera, HDMI output, dual microphones and support for 720p HD video. The GXV3350 adds dual Gigabit ports with PoE/PoE+, integrated Wi-Fi and Bluetooth support.

Protocols/Standards SIP RFC3261, TCP/IP/UDP, RTP/RTCP, HTTP/HTTPS, ARP, ICMP, DNS (A record, SRV, NAPTR), DHCP, PPPoE, SSH, TFTP, NTP, STUN, SIMPLE, LLDP-MED, LDAP, TR-069, 802.1x, TLS, SRTP, IPv6, OpenVPN®
Network Interfaces Dual switched 10/100/1000 Mbps ports with integrated PoE/PoE+
Camera Built-in 1 mega-pixel CMOS tiltable camera with privacy wheel, 720p 30fps
Bluetooth, Wi-Fi (2.4GHz & 5GHz)
Auxiliary Ports RJ9 headset jack (allowing EHS with Plantronics headsets), extension module port, USB port, HDMI-out (1.4 up to 720p30fps)
Voice Codecs and Capabilities Wide-band Opus, wide-band G.722, G.711µ/a, G. 729A/B, G.726-32, iLBC, in-band and out-ofband DTMF (In audio, RFC2833, SIP INFO), VAD, CNG, AEC, PLC, AJB, AGC, ANS
Video Codecs and Capabilities H.264 BP/MP/HP, video resolution up to 720p, frame rate up to 30 fps, bit rate up to 2Mbps, 3-way video conference (720p@30fps), anti-flickering, auto focus and auto exposure

GVC3212 provides up to 1080pHD video and comes equipped with integrated dual microphones that offer high quality voice pickup at up to 3-meter distance, advanced echo cancellation, and sophisticated background noise suppression. It supports Miracast and Airplay for convenient wireless content screen sharing, allowing meeting participants to share presentations, videos, or other content directly from their PC/ Mac or Android/iOS devices without tangling cables.

Platform IPVideoTalk
Network Interface 1x auto-sensing RJ45 10/100 Mbps Ethernet port
Display 1x HDMI 1.4 with support for up to 1080p video display
Camera Megapixel CMOS sensor, 720P@30fps
Lens 60 ° field-of-view wide angle
Remote Control Companion IR remote control
Wi-Fi Integrated dual-band Wi-Fi 802.11 a/b/g/n/ac
Auxiliary Ports TRS 3.5mm line in, 2x USB 2.0
Audio Codecs Full-band Opus, wide-band G.722, G.711, AEC, ANS, AGC, Noise Shield, PLC, CNG/VAD
Video Codecs and Capabilities H.264 BP/MP/HP, video resolution up to 720P 30fps; Content resolution up to 720P and up to 5fps; BFCP; anti-flickering, auto focus and auto exposure

GSC3510 robust SIP intercom device offers 2-way voice functionality with both a high-fidelity 8W HD speaker and 3 directional microphones with Multichannel Microphone Array Design (MMAD) that offer a 4.2 meter pickup distance. The GSC3510 supports a wide-range of peripherals including Bluetooth devices, built-in whitelist and blacklists to easily block unwanted calls, integrated dual-band Wi-Fi and advanced acoustic echo cancellation.

Protocols/Standards SIP RFC3261, TCP/IP/UDP, RTP/RTCP, HTTP/HTTPS, ARP, ICMP, DNS (A record, SRV, NAPTR), DHCP, PPPoE, SSH, TFTP, NTP, STUN, LLDP-MED, TR-069, 802.1x, TLS, SRTP, IPv6, OpenVPN®
Network Interfaces One 10/100 Mbps port with integrated PoE/PoE+
Bluetooth, Wi-Fi (2.4 & 5GHz with 802.11 a/b/g/n, WMM)
Auxiliary Port One 2-pin multi-purpose input port, Reset
Voice Codecs and Capabilities G.711µ/a, G.722 (wide-band), G.726-32, iLBC, Opus, G.729A/B in-band and out-ofband DTMF (In audio, RFC2833, SIP INFO), VAD, CNG, AEC, PLC, AJB, AGC, ANS
Telephony Features SIP Paging, Multicast Paging, call-waiting with priority override

GRP2615 is a high-end carrier-grade IP phone featuring a sleek design and a suite of next-generation features including integrated Wi-Fi, Bluetooth support, 40 multipurpose keys (MPKs), an available extension module, dual Gigabit ports and more.

Protocols/Standards SIP RFC3261, TCP/IP/UDP, RTP/RTCP, HTTP/HTTPS, ARP, ICMP, DNS(A record, SRV, NAPTR), DHCP, PPPoE, TELNET, TFTP, NTP, STUN, SIMPLE, LLDP, LDAP, TR-069, 802.1x, TLS, SRTP, IPV6
Network Interfaces Dual switched auto-sensing 10/100/1000 Mbps Gigabit Ethernet ports with integrated PoE
Bluetooth, Wi-Fi (2.4GHz & 5GHz)
Auxiliary Ports RJ9 headset jack (allowing EHS with Plantronics headsets), USB
Voice Codecs and Capabilities Support for G7.29A/B, G.711µ/a-law, G.726, G.722(wide-band), G723, iLBC, OPUS, in-band and out-of-band DTMF(in audio, RFC2833, SIP INFO), VAD, CNG, AEC, PLC, AJB, AGC

GRP2612 features a sleek design and a suite of next-generation features including Wi-Fi support. Key features include 16 virtual multi-purpose keys (VPKs), a color LCD with swappable face plates for easy logo customization and more.

Protocols/Standards SIP RFC3261, TCP/IP/UDP, RTP/RTCP, HTTP/HTTPS, ARP, ICMP, DNS(A record, SRV, NAPTR), DHCP, PPPoE, TELNET, TFTP, NTP, STUN, SIMPLE, LLDP, LDAP, TR-069, 802.1x, TLS, SRTP, IPV6
Network Interfaces Dual switched auto-sensing 10/100 Mbps Ethernet ports with integrated PoE
Wi-Fi (2.4GHz & 5GHz)
Auxiliary Ports RJ9 headset jack (allowing EHS with Plantronics headsets)
Voice Codecs and Capabilities Support for G7.29A/B, G.711µ/a-law, G.726, G.722(wide-band), G723, iLBC, OPUS, in-band and out-of-band DTMF(in audio, RFC2833, SIP INFO), VAD, CNG, AEC, PLC, AJB, AGC

GBX20 is an Extension Module that adds functionality, versatility and flexibility to the GRP2615, GRP2624, GRP2650, GRP2670 carrier-grade IP phones as well as and GXV3350 and GXV3450 IP video phones. It features a 272×480 LCD display that offers up to 40 contacts per module with support for up to 160 contact s by using 4 modules. The GBX20 supports a wide-range of advanced call features including BLF, call park /pick-up, speed-dial, presence, intercom, voice conferencing transfer/forward and much more.

Lines 20 per page (each module contains 2 pages, for up to 40 lines per module, Up to 160 with 4 daisy-chained modules
Compatible Grandstream IP phones GRP2615, GRP2624, GRP2650, GRP2670, GXV3350 and GXV3450
Feature Support Local GUI with animation driven from the host GRP2615 or GXV3350 phone; Multiple line/call appearances
Powered A single GBX20 can be powered by host phone (GRP2615 or GXV3350); when 2 or more GBX20 is connected an included 12V PSU is needed.

GAC2500 is designed to transform your business conferencing experience through its immersive audio conferencing environment and suite of advanced features. Redefining audio conferencing, the GAC’s 6 lines, WiFi and Bluetooth compatibility, HD Audio, Gigabit network ports and Android operating system with touch screen enhances the way users will meet and work together. Easily connect any time, using the GAC’s 7-way conference bridge. the GAC supports a productive experience with the GAC2500’s call scheduler, AndroidTM operating system and Google Play Store access.

Protocols/Standards SIP RFC3261, TCP/IP/UDP, RTP/RTCP, HTTP/HTTPS, ARP, ICMP, DNS (A record, SRV, NAPTR), DHCP, PPPoE, SSH, TFTP, NTP, STUN, SIMPLE, LLDP, LDAP, TR-069, 802.1x,
TLS, SRTP, IPV6 (pending), OpenVPN (pending)
Network Interfaces Auto-sensing Gigabit Ethernet port with integrated PoE+ 
Bluetooth, Wi-Fi
Auxiliary Ports 3.5mm audio port, USB Micro-B, RJ48 daisy chain port
Voice Codecs and Capabilities Support for G.711µ/a, G.722, G.726, iLBC, Opus, G.722.1 and G.722.1c (pending), in-band and out-of-band DTMF (In audio, RFC2833, SIP INFO), G.729A/B, VAD, CNG,
AEC, PLC, AJB, AGC

DP760 is a powerful wideband DECT repeater (wireless relay station) that auto associates to Grandstream’s DP750/DP752 DECT base stations, offering extended mobility to business and residential users. The DP760 extends an additional range of 300 meters outdoors and 50 meters indoors to give users the freedom to move around their home or work space. This Wideband DECT Repeater relays up to 2 concurrent HD calls. The Ethernet connection provides PoE for convenient installation and a variety of remote features including provisioning, status monitoring, and repeater firmware upgrades.

Protocols/Standards TCP/IP/UDP, HTTP/HTTPS, ARP/RARP, ICMP, DNS, DHCP, PPPoE, SSH, TFTP, NTP, LLDP-MED, UPnP
Voice Codecs and Capabilities G.722 codec for HD audio and G.726 codec for narrow band audio
Automatic or manual association to DP750/DP752, base stations for easy use

HT802 delivers powerful VoIP technology and routing capabilities to home and office environments, and allows users to successfully connect their analog devices to a manageable and powerful VoIP network. Built upon Grandstream’s market-leading SIP ATA/gateway technology, with millions of units successfully deployed worldwide, this powerful ATA features exceptional voice quality in various applications and environments.

FXS Ports 2
Network Interfaces 1 Fast Ethernet auto-sensing port
Voice Codecs G.711 with Annex I (PLC) and Annex II (VAD/CNG), G.722, G.723.1, G.729A/B, G.726, iLBC, OPUS, dynamic jitter buffer, advanced line echo cancellation
Fax Over IP T.38 compliant Group 3 Fax Relay up to 14.4kpbs and auto-switch to G.711 for Fax Pass-through
Provisioning and Control HTTP, HTTPS, SSH, TFTP, TR-069, secure and automated provisioning using AES encryption, syslog
Universal Power Supply Input: 100-240VAC, 50-60Hz, Output: 5.0VDC/1.0A

GXP1625 is a reliable Basic IP phone for the user who requires standard features for a light to medium call volume. Stay in touch with others with its 2 lines/SIP accounts, enjoy crystal clear HD audio and utilize its dual-switched 10/100 mbps ports with integrated PoE for a flexible deployment. Maximize your productivity with essential supported features such as 3-way conferencing and 3 XML programmable soft keys. Together, these featuresmake the GXP1620/25 an easy-to-use and effective Basic IP phone.

Protocols/Standards SIP RFC3261, TCP/IP/UDP, RTP/RTCP, HTTP/HTTPS, ARP/RARP, ICMP, DNS (A re- cord, SRV, NAPTR), DHCP, PPPoE, SSH, TFTP, NTP, STUN, SIMPLE, LLDP-MED, LDAP,
TR-069, 802.1x, TLS, SRTP, CDP/SNMP/RTCP-XR
Network Interfaces Dual switched auto-sensing 10/100 Mbps Ethernet ports, integrated PoE
Auxiliary Ports RJ9 headset jack (allowing EHS with Plantronics headsets), USB, extension module port
Voice Codecs and Capabilities Support for G.711µ/a, G.722 (wide-band), G.723, G.726-32, G.729 A/B, iLBC, inband and out-of-band DTMF (In audio, RFC2833, SIP INFO), VAD, CNG, AEC, PLC,
AJB, AGC