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Grandstream DP710 Cordless Additional IP Phone for DP715 phones and VoIP devices

Grandstream

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Up to 5 DP710 phones can be registered to the DP715 base station.
Security protection: SIP via TLS and SRTP
Battery life: 10 hours talk time, 80 hours standby

  Grandstream DECT IP Phones Datasheet (PDF)

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Grandstream DP710 Cordless Additional IP Phone for DP715 phones and VoIP devices

Grandstream DP710 is the next-generation high-performance, economical, high-quality and easy-to-set DECT VoIP phone. It is  suitable for small businesses and home users. Its compact size, excellent voice quality, wide range of features, market-leading prices and wide wireless coverage enables consumers to maximize the performance and portability of voice over IP applications with minimal investment. DP715 / 710 is compatible with SIP and DECT.

DECT base station (included in DP715) can register up to 5 DECT phones and communicate with 4 phones at the same time. Secure via HTTP / HTTPS / Telnet / TFTP And automatically provide SIP account, SIP over TCP / TLS, SRTP
including billing support

Grandstream DP710 Cordless Additional IP Phone has extended telephone functions, such as caller ID, call waiting, 3-party conference, forwarding, do not disturb. Additionally, other features includes: waiting message display (intermittent tone ), automatic answer, multi-language voice prompts, flexible dial plan etc.

Grandstream DP710 Cordless Additional IP Phone

Grandstream DP710 Additional Handset Technical Specification

Air Interfaces

  • Telephony standards: DECT / GAP Frequency range: 1880 – 1900 MHz (Europe), 1920 – 1930 MHz (US) Number of channels: 120 (Europe), 60 duplex (US) Modulation: GFSK Speech coding: 32 kbit/s Emission power: 10 mW (average power per channel) Range: up to 300 m outdoors, maximum of 50 m in buildings

Handset Display

  • 1.7″ 102×80 FSTN LCD with color backlight

Audio Interface

  • Handsfree speaker

Telephony Features

  • Caller ID display or block, call waiting, Flash, blind or attended transfer, forward, hold, do not disturb, 3-way conference

QoS

  • Layer 2 (802.1Q VLAN/802.1p), Layer 3 (ToS, DiffServ, MPLS)

IP Transport

  • RTP/RTCP

DTMF Method

  • In-audio, RFC2833 and/or SIP Info

IP Signaling

  • SIP (RFC 3261)

Multiple SIP accounts per base station

  • Up to five (5) distinct SIP accounts per system; Independent SIP account per handset; Multiple handsets per SIP account

Hunting Group

  • Linear mode; Parallel mode; Shared Line mode

Provisioning

  • HTTP, HTTPS, TELNET, TFTP, TR-069 (pending), secure and automated provisioning

Security

  • Security protection: SIP over TLS and SRTP.

Device Management

  • Web interface or secure (AES encrypted) central configuration file for mass deployment

Phonebook(Per Handset)

  • 200 numbers (up to 24 digits) with an associated name (up to 16 characters); 10 outgoing call entries; 30 incoming calls entries

Multi-language Input

  • English; Latin; Greek; Russian

Polyphonic Ringtones

  • 18 different ringer melodies are available to indicate an incoming call (internal intercom or external VoIP)

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