Product Overview; Software Features Overview; Table 5: Dp715/Dp710Technical Specifications - Grandstream Networks DP715 User Manual

The DP715/710 is the next generation of powerful, affordable, high quality and simple to configure DECT
Cordless IP Phone for small business and residential users. Their compact size, superb voice quality, rich
feature set, market leading price-performance and wide range radio coverage enable consumers to
maximize the power of IP voice application and mobility for a minimum investment. The VoIP network
signaling protocol supported is SIP. The DP715/710 is fully compatible with SIP industry standard and can
interoperate with many other SIP compliant devices and software on the market. Moreover, it supports
comprehensive voice codecs including G.711, G.723.1, G.729AB, G.726 and iLBC.

SOFTWARE FEATURES OVERVIEW

Air Interfaces
Network Interface
LED Indicators
Handset Display
Factory Reset Button
Audio Interface
Voice over Packet
Capabilities
Voice Compression
Telephony Features
QoS
IP Transport
DTMF Method
IP Signaling
Multiple SIP accounts
per base station
Hunting Group
Provisioning
Security
Device Management
Firmware version 1.0.0.31

PRODUCT OVERVIEW

Table 5: DP715/DP710TECHNICAL SPECIFICATIONS

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
One 10/100Mbps auto-sensing Ethernet port (RJ45) ( DP715 Base Station only)
Base Station : Power, Network, Register, Call
1.7" 102x80 FSTN LCD with color backlight
Yes ( DP715 Base Station only)
Handsfree speaker (Handset only)
Base Station : Dynamic Jitter Buffer
Handset : Speakerphone with Acoustic Echo Cancellation
G.711 with Annex I (PLC) and Annex II (VAD/CNG), G.723.1, G.726-32 AAL2,
G.729A/B, iLBC
Caller ID display or block, call waiting, Flash, blind or attended transfer, forward, hold, do
not disturb, 3-way conference
Layer 2 (802.1Q VLAN/802.1p), Layer 3 (ToS, DiffServ, MPLS)
RTP/RTCP
In-audio, RFC2833 and/or SIP Info
SIP (RFC 3261)
Up to five (5) distinct SIP accounts per system; Independent SIP account per handset;
Multiple handsets per SIP account
Linear mode; Parallel mode; Shared Line mode
HTTP, HTTPS, TELNET, TFTP, TR-069 (pending), secure and automated provisioning
Security protection: SIP over TLS and SRTP.
Web interface or secure (AES encrypted) central configuration file for mass deployment
Support device configuration via built-in IVR, Web browser or central configuration file
through TFTP, HTTP or HTTPS
Auto/manual provisioning system
NAT-friendly remote software upgrade for deployed devices including behind
firewall/NAT
Syslog support
DP715/DP710 User Manual Page 13 of 56