Internet Telephone
24-Port VoIP Station Gateway

ƒ24 high-density FXS ports

ƒIdeal for business phones/fax

ƒCompatible with SIP services

ƒExtensive call feature support

ƒGuaranteed toll-quality voice even on
busy networks
Call Features

ƒCall hold, wait, forward, transfer
A Comprehensive VoIP Solution
Guaranteed Voice Quality

ƒEasy tracking with call detail record
The DVG-2024S VoIP Station Gateway presents an The DVG-2024S gateway delivers clear voice

ƒGreeting message
ideal Internet telephone solution for business use. and reliable phone/fax communication through
This gateway converts voice traffic into data packets implementation of internationally recognized

ƒCaller ID
for transmission over the Internet. It combines standards for voice and data networking. It
WAN/LAN Connection
the industry’s latest Voice over IP (VoIP) network incorporates Quality of Service (QoS) functions

ƒ10/100 Mbps Ethernet WAN Port
technology with advanced communication features, to ensure that audio quality received through the
and is fully compatible with SIP Internet phone Internet is the same as or even surpasses that

ƒ10/100 Mbps Ethernet LAN Port
services. High port densities allow it to provide a low received on a standard land line.

ƒWAN Connection supports PPPoE,
cost of ownership, convenience, and great savings
for companies needing to place frequent long-
Convenient Call Features
distance and international business calls.
The DVG-2024S gateway supports extensive call
features such as call waiting and call forwarding,

ƒEasy Configuration by IVR or Web-
Cost Saving and Investment Protection
allowing service providers to offer these functions
Based GUI
The DVG-2024S gateway provides businesses with an to all subscribers with compatible telephones.
easy and inexpensive upgrade to Internet telephony Configuration of an individual phone connection

ƒWeb-Based Firmware Upgrade
while allowing them to retain their existing telephone is easy using the multi-language Interactive Voice
and fax equipment. These devices allow businesses Response (IVR) system or the web-based user
to protect and extend their past investments in interface.
telephones, conference speakers phones and fax
machines, as well as to control their migration to VoIP
with a very affordable and incremental investment.
24 Internet Telephone Connections
This gateway provides 24 FXS ports for simultaneous
Internet phone connections. Plug in regular phone
sets to these ports and they instantly become
Internet telephones. For businesses with a frequent
need for long-distance and international phone
calls, the VoIP gateway provides great cost savings
and convenience while keeping configuration and
maintenance to a minimum.

24-Port VoIP Station Gateway
Technical Specifications
Voice Features
- Loopback – analogue

ƒ G.711 a/μ-law, G.729A/B, G.723.1, G.726
- SLIC DC power voltage
- Tip / Ring DC feed
Packet Interval (ms)
Concurrent Call
- Ringer
-Outward Test (GR909 Standard) :
G.711 a-law
20 ms, 30 ms, 40 ms
24 ch @ 20 ms
G.711 u-law
20 ms, 30 ms, 40 ms
24 ch @ 20 ms
- Phone Line disconnected
- H.F. DC Voltage (Hazardous and foreign DC Voltage)
20 ms, 30 ms, 40 ms
24 ch @ 20 ms
- H.F. AC Voltage (Hazardous and foreign AC Voltage)
30 ms, 60 ms, 90 ms
24 ch @ 30 ms
- Tip / Ring Short

ƒ Modem over IP up to V.34
20 ms, 30 ms, 40 ms
24 ch @ 20 ms

ƒ ROH Tone (Receiver Off-Hook Tone @ 480 Hz)

ƒ DTMF Detection and Generation

ƒ Loop Current Suppression

ƒ Silence Suppression & Detection
SIP Account Management

ƒ Comfort Noise Generation (CNG)

ƒ By Port Registration

ƒ Voice Activity Detection (VAD)

ƒ By Device Registration (share account)

ƒ Echo Cancellation (G.165/G.168)

ƒ Mixed Mode (Hunt number for inbound, by port number for outbound)

ƒ Adaptive (Dynamic) Jitter Buffer

ƒ Invite with Challenge

ƒ Call Progress Generation

ƒ Register by SIP Server IP Address or Domain Name

ƒ Auto or Programmable Gain Control

ƒ Support RFC3986 SIP URI Format

ƒ ITU-T V.152 Voice-band Data over IP Networks
SIP Call Management
SIP Call Features

ƒ Support Outbound Proxy

ƒ Peer to Peer Call

ƒ Register up to three SIP servers

ƒ Call Hold / Retrieve

ƒ SIP Registration Failover Mechanism

ƒ Call Waiting

ƒ Group Hunting

ƒ Call Pick Up

ƒ Privacy Mechanism / Private Extensions to SIP

ƒ Call Park / Retrieve (SIP Server Required)

ƒ Session Timers (Update / Re-invite)

ƒ Call Forward - unconditional, busy, no answer

ƒ DNS SRV Support

ƒ Call Transfer - attended, unattended

ƒ Call Types: Voice / Modem / FAX

ƒ Do Not Disturb

ƒ Call Routing by Prefix Number

ƒ Speed Dialing

ƒ User Programmable Dial Plan Support

ƒ Repeat Dialing

ƒ CDR Client
ƒ Three-way Calling

ƒ Manual Peer Table (for P2P calls)
ƒ MWI (RFC-3842)

ƒ E.164 Numbering, ENUM support
ƒ Hot Line and Warm Line
IP Network Specifications
Telephony Specifications

ƒ Support IPv4, IPv6 future upgradable (Option)
ƒ In-Band DTMF, Out-of-Band DTMF Relay (RFC2833 or SIP INFO)

ƒ DTMF / PULSE Dial Support

ƒ Network Protocol Support:
ƒ Caller ID Generation / Detection:

-FSK-Bellcore Type 1 & 2

ƒ HTTP, HTTPS, DNS, DNS SRV, Telnet, DHCP Server, DHCP Client, STUN Client, UPnP,
-FSK-ETSI Type 1 & 2

ƒ QoS Support:
-WAN: DiffServ, IP Precedence, Priority Queue, Rate Control, 802.1Q (VLAN Tagging),
-FSK: Calling Name, Number, Date and Time, VMWI
802.1p (Priority Tag)

ƒ FXS Metering Pulse:

ƒ DDNS Support
-Polarity Reversal
Network Security Specifications
-12kHz calling tone
-16kHz calling tone

ƒ VPN PPTP Client

ƒ T.30 FAX Bypass to G.711, T.38 Real Time FAX Relay

ƒ DIGEST Authentication

ƒ FXS Line test and diagnostics with visual alarm indication

ƒ MD5 Encryption
-Inward self test:

ƒ DoS Protection
- Loopback – codec

24-Port VoIP Station Gateway

ƒ RFC1035 Domain Names - implementation and specification

ƒ Web-Based Configuration

ƒ RFC1058 Routing Information Protocol (RIP)

ƒ Auto-provisioning (HTTP / HTTPS)

ƒ RFC1157 Simple Network Management Protocol (SNMP)

ƒ Telnet

ƒ RFC1305 Network Time Protocol (NTP)


ƒ RFC1321 The MD5 Message-Digest Algorithm

ƒ FTP / TFTP / HTTP Software Upgrade

ƒ RFC1349 Type of Service in the Internet Protocol Suite

ƒ Configuration Backup and Restore

ƒ RFC1350 TFTP Protocol (Revision 2)

ƒ Reset to Default Button

ƒ RFC1661 Point-to-Point Protocol (PPP)

ƒ TR-069/104 (Option)

ƒ RFC1738 Uniform Resource Locators (URL)

ƒ RFC2854 The ‘text/html’ Media Type
SIP, Voice and FAX Related Standard

ƒ RFC2131 Dynamic Host Configuration Protocol (DHCP)

ƒ RFC1889 RTP: A Transport Protocol for Real-Time Applications.

ƒ RFC2136 Dynamic Updates in the Domain Name System (DNS UPDATE)

ƒ RFC2543 SIP: Session Initiation Protocol

ƒ RFC2327 SDP: Session Description Protocol

ƒ RFC2833 RTP Payload for DTMF Digits, Telephony Tones and Telephony Signals

ƒ RFC2474 Definition of the Differentiated Services Field (DS Field)

ƒ RFC2880 Internet Fax T.30 Feature Mapping

ƒ RFC2516 A Method for Transmitting PPP Over Ethernet

ƒ RFC2976 The SIP INFO Method

ƒ RFC2616 Hypertext Transfer Protocol - HTTP/1.1

ƒ RFC3261 SIP: Session Initiation Protocol

ƒ RFC2617 HTTP Authentication: Basic and Digest Access Authentication

ƒ RFC3262 Reliability of Provisional Responses in Session Initiation Protocol (SIP)

ƒ RFC2637 Point-to-Point Tunneling Protocol

ƒ RFC3263 Session Initiation Protocol (SIP): Locating SIP Servers

ƒ RFC2766 Network Address Translation - Protocol Translation (NAT-PT)

ƒ RFC3264 An Offer/Answer Model with Session Description Protocol (SDP)

ƒ RFC2782 A DNS RR for Specifying the location of Services (DNS SRV)

ƒ RFC3265 Session Initiation Protocol (SIP) - Specific Event Notification


ƒ RFC3311 The Session Initiation Protocol (SIP) UPDATE Method

ƒ RFC2916 E.164 Number and DNS

ƒ RFC3323 A Privacy Mechanism for the Session Initiation Protocol (SIP)

ƒ RFC3022 Traditional IP Network Address Translator

ƒ RFC3325 Private Extensions to the Session Initiation Protocol (SIP) for Asserted
Identity within Trusted Networks

ƒ RFC3489 STUN - Simple Traversal of User Datagram Protocol (UDP) through Network
Address Translators (NATs)

ƒ RFC3362 Real-time Facsimile (T.38) - Image/t38 MIME Sub-type Registration

ƒ RFC3515 The Session Initiation Protocol (SIP) Refer Method
Power Input

ƒ RFC3550 RTP: A Transport Protocol for Real-Time Applications. July 2003

ƒ 100 to 240 V AC

ƒ RFC3665 Session Initiation Protocol (SIP) Basic Call Flow Examples

ƒ Power consumption: 70 W

ƒ RFC3824 Using E.164 numbers with the Session Initiation Protocol (SIP)

ƒ MTBF: 78864 hrs

ƒ RFC3841 Caller Preferences for the Session Initiation Protocol (SIP)

ƒ RFC3842 A Message Summary and Message Waiting Indication Event Package for the
Session Initiation Protocol (SIP)

ƒ 445 x 240 x 45 mm (17.5 x 9.4 x 1.8 inches)

ƒ RFC3891 The Session Initiation Protocol (SIP) “Replaces” Header
ƒ 19-inch standard rack-mount width, 1U height

ƒ RFC3892 The Session Initiation Protocol (SIP) Referred-By Mechanism

ƒ RFC3960 Early Media and Ringing Tone Generation in the Session Initiation Protocol

ƒ 3.4 kg (7.5 lb)

ƒ RFC3986 Uniform Resource Identifier (URI): Generic Syntax
Operating Temperature

ƒ RFC4028 Session Timers in the Session Initiation Protocol (SIP)

ƒ 0° to 40° C (32° to 104° F)

ƒ Draft-ietf-sipping-service-examples-08 for call features
Storage Temperature
Network Related Standards

ƒ -20° to 60° C (-4° to 140° F)

ƒ RFC318 Telnet Protocols

ƒ RFC791 Internet Protocol

ƒ Up to 95% non-condensing

ƒ RFC792 Internet Control Message Protocol (ICMP)

ƒ RFC793 Transmission Control Protocol (TCP)

ƒ RFC768 User Datagram Protocol (UDP)

ƒ FCC Class A

ƒ RFC826 Ethernet Address Resolution Protocol (ARP)

ƒ CE

ƒ RFC959 File Transfer Protocol (FTP)


ƒ RFC1034 Domain Names - concepts and facilities
D-Link Corporation
No. 289 Xinhu 3rd Road, Neihu, Taipei 114, Taiwan
Specifications are subject to change without notice.
D-Link is a registered trademark of D-Link Corporation and its overseas subsidiaries.
All other trademarks belong to their respective owners.
©2010 D-Link Corporation. All rights reserved.
Release 01 (August 2010)