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Top » Catalog » VoIP / SIP Phones » IP Desk Phones » Standard » GXP1628


Grandstream GXP1628 HD IP Phone
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Grandstream GXP1628 HD IP Phone
[GXP1628]
  • Next-generation 2-line enterprise/business SIP phone
  • HD audio quality and rich feature functionality
  • Dual RJ-45 10/100/1000Mbps ports with integrated PoE
  • 8 XML speed dial / BLF keys and 5 XML context sensitive soft keys
  • HD audio handsfree speaker phone with echo cancellation
  • 2 year warranty against hardware defects
  • 14 day money back guarantee
  • 30 day email support for technical installation issues
Product Overview
The Grandstream GXP1628 is a powerful IP phone for small-to-medium businesses (SMBs). This Linux-based model includes 2 lines, 3 XML programmable soft keys, 8 BLF keys and 3-way conferencing. A 132x48 backlit LCD screen creates a clear display for easy viewing. Additional features such as dual switched gigabit network ports, HD audio, multi-language support, integrated PoE, and call-waiting allow the GXP1628 to be a high quality, versatile and dependable office phone.

Feature Summary
  • HD wideband audio handset
  • Full-duplex hands-free speakerphone with advanced acoustic echo cancellation
  • Automated phone book synchronization with directory server using XML. Large phonebook (up to 500 contacts) and call history (up to 200 records)
  • Integrated real-time web applications (weather, stock, currency, RSS news, etc.)
  • 2 dual-colour line keys with dual colour LED line indicators and 2 independent SIP accounts, 8 BLF keys, 3 XML dynamic context sensitive soft keys, and 5 navigation/menu/volume keys
  • 13 dedicated function keys for MUTE, HEADSET, TRANSFER, CONFERENCE, SEND and REDIAL, SPEAKERPHONE, VOLUME, PHONEBOOK, MESSAGE, HOLD, PAGE/INTERCOM, RECORD, HOME
  • Voice Features: Downloadable Phone Book (XML, LDAP), XML Customisation of Screen, Voice Mail Indicator, Redial, Call Log, Volume Control, Caller ID Display or Block, Call Waiting, Hold, Transfer, Forward, FLASH, Mute, 3-Way Conferencing, off-hook auto dial, configurable emergency dialling (e.g., 911), early dial, click-to-dial, auto answer, downloadable ring tones
  • Dual RJ-45 10/100/1000 Gigabit autosensing ports (switched or routed) to allow a desk PC to be connected to the phone in order to conserve switch/router ports and AC sockets in order to simplify deployment
  • Integrated PoE: IEEE 802.3af Class 2
  • RJ9 headset jack (allowing EHS with Plantronics headsets)
  • NAT router and DHCP server
  • Multi language support (English, German, Italian, French, Spanish, etc.)
  • Secure Real-time Transport Protocol (SRTP) and SIP Transport Layer Security (TLS) (pending) support
  • Full duplex handsfree speakerphone with Advanced Echo Cancellation (AEC), Acoustic Gain Control (AGC) and side tone support
  • Easy configuration through manual operation (phone keypad and Web interface) or personalized automated provisioning via central configuration file for mass deployment.
  • NAT-friendly remote firmware upgrade capability via tftp/http even from behind firewalls/NATs
  • Interoperable with various 3rd party SIP end user device, Proxy/Registrar/Server, and gateway products (e.g. 3CX, Asterisk®/Trixbox® IP PBX, MS Messenger, Cisco® IP phone and gateway, etc)
  • Dynamic negotiation of codec and voice payload length
  • SIP and DNS server redundancy and failover
  • Features Advanced Digital Signal Processing (DSP) to ensure superb hi-fidelity audio quality
  • Advanced and patent pending adaptive jitter buffer control, packet delay and loss concealment technology
Technical Summary
  • SIP Presence package (RFC3856, 3863) for use of 2 programmable line keys, SIP MESSAGE method (RFC3428) for up to 100 incoming IM messages, SIP PUBLISH method (RFC3906), SIP dialogue package (RFC4235)
  • Voice codecs supported include G.723.1 (5.3K/6.3K), G.729A/B, G.711 (a-law and u-law), G.722 (wide-band), G.726 (32K), GSM and iLEB codecs.
  • Silence Suppression, VAD (Voice Activity Detection), CNG (Comfort Noise Generation), Line Echo Cancellation (G.168), and AGC (Automatic Gain Control)
  • In-band and out-of-band DTMF (in audio, RFC2833, SIP INFO)
  • Support for Fixed IP, DHCP, SIP Presence (SIMPLE), Automated NAT traversal using IETF STUN (manual configuration of firewall/NAT not required) and symmetric RTP (compatible with Cisco’s ATA-186, etc)
  • Protocols supported include SIP RFC3261, TCP/UDP/IP, RTP/RTCP, HTTP, ARP/RARP, ICMP, DNS (A record and SRV), DHCP (client and server), NTP, TFTP, PPPoE
  • Support for Layer 2 (802.1Q VLAN, 802.1p) and Layer 3 QoS (ToS, DiffServ, MPLS)
  • DIGEST authentication and encryption using MD5 and MD5-sess
Hardware Summary
  • LAN interface: 2 x RJ45 10/100/1000Mbps autosensing, PoE: IEEE 802.3af Class 2
  • Headset Jack: RJ9
  • LED: 2 dual colour line indicator LEDs and 1 message indicator LED
  • Display: 132x48 pixel graphical LCD display
  • Universal Switching Power Adapter: Input: 100-240VAC; 50-60 Hz, Output: +5VDC, 600mA
  • Dimensions: 220.5mm (L) x 192.5mm (W) x 76.0mm (H) (with handset)
  • Handset Weight: 0.9KG
  • Package Weight: 1.2KG
  • Temperature: 32 - 104 ºC (0 - 40 ºF)
  • Humidity 10-90% (non condensing)
  • FCC: Part 15 (CFR 47) Class B, CE : EN55022 Class B, EN55024, EN61000-3-2, EN61000-3-3, EN60950-1RCM: AS/ACIF S004; AS/NZS CISPR22/24; AS/NZS 60950; AS/NZS 60950.1


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This product was added to our catalog on Tuesday 24 August, 2010.
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