DVG-5008SGמיועד לפרויקטים
תאור
DVG-5008SG converts voice traffic into data packets for transmission over the Internet. The Gateway combines the latest Voice over IP technology with advanced communication features, and is fully compatible with Internet-phone SIP. Gateways with high density and low cost, provide convenience and guaranteed savings for companies needing to place frequent long-distance and international business calls.
Saving and a profitable investment
VoIP-Gateway DVG-5008SG with FXS-port provides an easy and inexpensive upgrade Internet telephony, allowing users to store previously purchased phones and fax machines. Investment protection is achieved through the use of existing infrastructure and the possibility of its gradual modernization.
Guaranteed quality of voice
Gateway DVG-5008SG transmits voice and fax messages in accordance with internationally recognized standards for voice and data, such as G.722, G.711, etc. Support for Quality of Service (QoS) ensures call quality comparable to analog phone.
Call Functions
VoIP-Gateway DVG-5008SG with FXS-port supports multiple calling features that allow service providers to provide their customers with services such as call waiting, "Do Not Disturb", speed dialing, three-way conference, etc. Besides, there are many management functions SIP, including Proxy server support for outgoing calls, duplicate registration for SIP proxy servers / failover registration SIP, function dial plan (dial plan) and the group call function. Customize the phone connection by running the multi interactive voice (IVR) or the Web-interface.
Multiple simultaneous connections
VoIP-Gateway DVG-5008S is equipped with eight FXS-ports, providing multiple simultaneous connections. For the organization of Internet phone just connect to these ports regular phones. These gateways are the ideal solution for companies to make phone calls to long-distance and international destinations.
Gigabit ports to provide broadband Internet access
Gateway provides easy routing function, which allows users to get a joint office broadband Internet access. You can connect your computer to this gateway with built-in 4-port Ethernet and use the built-in NAT / DHCP-server to automatically access the Internet.
מאפיינים כללים
Physical interfaces
• WAN: 1 port 10/100/1000 Mb / s Ethernet
• LAN: 4 x 10/100/1000 Mbit / s Ethernet
• Ports Phone: 8-port FXS (RJ-11)
• Button Reset: Returns to the default settings
• Button Power: on / off switch power
Software features
Voice Features
• G.722, G.711 a / u-law, G.723.1, G.726, G.729A / B, GSM 6.10 Full Rate, iLBC13, 3 kbit / s
• DTMF detection and generation
• Detect and silence suppression
• Comfort Noise Generation (CNG)
• Voice Activity Detection (VAD)
• Echo cancellation (G.165/G.168)
• Variable (dynamic) jitter-buffer
• Generate tone CPT (Call Progress Tone)
• Automatic or programmable gain
• Integrated local frequency converter
• Supports ITU-T V.152
Phone functions
• In-Band DTMF, Out-of-Band DTMF Relay (RFC2833 and SIP INFO)
• Support for DTMF / Pulse
• Detection / Generation Caller ID (Caller ID):
- DTMF
- FSK-Bellcore Type 1 and 2
- FSK-ETSI Type 1 and 2
- FSK-NTT
- FSK: caller's name, number, date and time, vMWI
• billed pulses at port FXS:
- Detection of reverse polarity
- Tonal range of 12 kHz
- Tonal range of 16 kHz
• Generation (FXS) tone PSTN
• T.30 FAX Bypass, T.38 Real Time FAX Relay
• Working modem connection via V.34
• Signal ROH (Receiver Off-Hook, @ 480 Hz)
• Loop Current Suppression
• Diagnostic and testing lines FXS with visual indication of errors
• Incoming calls:
- Loopback-codec
- Loopback-analogue
- Voltage SLIC
- Wire a / b powered by DC
- Incoming call
• Outgoing calls (Standard GR909):
- REN
- Disconnect
- Danger high voltage direct current
- Danger high voltage AC
- Wire a / b short
SIP-supported methods
• ACK
• BYE
• CANCEL
• INFO
• INVITE
• MESSAGE
• NOTIFY
• OPTIONS
• PING
• PRACK
• PUBLISH
• REFER
• REGISTER
SIP-calls
• Calls «Peer-to-Peer»
• Hold
• Call waiting / call return
• Call Pickup
• Parking / call return (requires SIP-server)
• Call forwarding (unconditional, busy, no answer)
• Call transfer (with and without announcement)
• "Do not disturb"
• Speed dialing
• Redial
• Three-way conference
• MWI (RFC-3842)
• Hot Line and Warm Line
Call Management
• Support for outbound proxy
• Up to three SIP-server
• Re-registration for SIP-server authorization fails
• The group call
• Privacy Mechanism / Private Extensions to SIP
• Session Timers (Update / Re-invite)
• Support for DNS SRV
• The types of calls: Voice / Modem / Fax
• Call Routing prefix
• Support for user-programmable dial plan
• CDR-client
• Create a manual recording (for calls P2P)
• Support for ENUM and E.164 numbering standard
Manage Accounts
• Registration for the port-based
• Registration for the core of the device (general account)
• Mixed mode register (Hunt number for incoming calls, based on the port number for outgoing calls)
• Invite with Challenge
• Registration for the IP-address or domain name server SIP-
• Supports RFC3986 SIP URI
IP specification
• Support for IPv4, IPv6
• WAN: Static IP, PPPoE, DHCP, PPTP
• Supports network protocols: IP, TCP, UDP, TFTP, FTP, RTP, RTCP, ARP, RARP, ICMP, NTP, SNTP, SNMP v1/v2/v3, HTTP, HTTPS, DNS, DNS SRV, Telnet, DHCP-server , DHCP-client, STUN-Client, UPnP, IGMP snooping, IGMP proxy
• Support for QoS:
- WAN: DiffServ, IP Precedence, priority queues, Rate Control, 802.1Q (VLAN Tagging), 802.1p (Priority Tag)
- LAN: Speed Limit
• Support DDNS
- Dyndns.org, TZO and Peanut Hull
Network Security
• VPN PPTP-client
• Authentication DIGEST
• MD5 Encryption
• Protection against DoS attacks
Management
• Web-interface
• Auto-provisioning (HTTP / HTTPS)
• Telnet
• IVR
• Software upgrade via FTP / TFTP / HTTP
• Backup and restore settings
• The Reset button to reset to the default settings
• TR-069/104 (optional)
• SNMP V3 / V2c / V1
Standards SIP, Voice and FAX
• RFC1889 RTP: A Transport Protocol for Real-Time Applications
• RFC2543 SIP: Session Initiation Protocol
• RFC2833 RTP Payload for DTMF Digits, Telephony Tones and Telephony Signals
• RFC2880 Internet Fax T.30 Feature Mapping
• RFC2976 The SIP INFO Method
• RFC3261 SIP: Session Initiation Protocol
• RFC3262 Reliability of Provisional Responses in Session Initiation Protocol (SIP)
• RFC3263 Session Initiation Protocol (SIP): Locating SIP Servers
• RFC3264 An Offer / Answer Model with Session Description Protocol (SDP)
• RFC3265 Session Initiation Protocol (SIP)-Specific Event Notification
• RFC3311 The Session Initiation Protocol (SIP) UPDATE Method
• RFC3323 A Privacy Mechanism for the Session Initiation Protocol (SIP)
• RFC3325 Private Extensions to the SessionInitiation Protocol (SIP) for Asserted Identity within Trusted Networks
• RFC3362 Real-time Facsimile (T.38) -image/t38 MIME Sub-type Registration
• RFC3515 The Session Initiation Protocol (SIP) Refer Method
• RFC3550 RTP: A Transport Protocol for Real-Time Applications. July 2003
• RFC3665 Session Initiation Protocol (SIP) Basic Call Flow Examples
• RFC3824 Using E.164 numbers with the Session Initiation Protocol (SIP)
• RFC3841 Caller Preferences for the Session Initiation Protocol (SIP)
• RFC3842 A Message Summary and Message Waiting Indication Event Package for the Session Initiation Protocol (SIP)
• RFC3891 The Session Initiation Protocol (SIP) "Replaces" Header
• RFC3892 The Session Initiation Protocol (SIP) Referred-By Mechanism
• RFC3960 Early Media and Ringing Tone Generation in the Session Initiation Protocol (SIP)
• RFC3986 Uniform Resource Identifier (URI): Generic Syntax
• RFC4028 Session Timers in the Session Initiation Protocol (SIP)
• Draft-ietf-sipping-service-examples-08 for call features
Network Standards
• RFC318 Telnet Protocols
• RFC791 Internet Protocol
• RFC792 Internet Control Message Protocol
• RFC793 Transmission Control Protocol
• RFC768 User Datagram Protocol
• RFC826 Ethernet Address Resolution Protocol
• RFC959 File Transfer Protocol
• RFC1034 Domain Names-concepts and facilities
• RFC1035 Domain Names-implementation and specification
• RFC1058 Routing Information Protocol
• RFC1157 Simple Network Management Protocol (SNMP)
• RFC1305 Network Time Protocol (NTP)
• RFC1321 The MD5 Message-Digest Algorithm
• RFC1349 Type of Service in the Internet Protocol Suite
• RFC1350 The TFTP Protocol (Revision 2)
• RFC1661 The Point-to-Point Protocol (PPP)
• RFC1738 Uniform Resource Locators (URL)
• RFC2854 The 'text / html' Media Type
• RFC2131 Dynamic Host Configuration Protocol
• RFC2136 Dynamic Updates in the Domain Name System (DNS UPDATE)
• RFC2327 SDP: Session Description Protocol
• RFC2474 Definition of the Differentiated Services Field (DS Field)
• RFC2516 A Method for Transmitting PPP Over Ethernet
• RFC2616 Hypertext Transfer Protocol -HTTP/1.1
• RFC2617 HTTP Authentication: Basic and Digest Access Authentication
• RFC2637 Point-to-Point Tunneling Protocol
• RFC2766 Network Address Translation-Protocol Translation (NAT-PT)
• FC2782 A DNS RR for Specifying the location of Sevices (DNS SRV)
• RFC2818 HTTP Over TLS (HTTPS)
• RFC2916 E.164 Number and DNS
• RFC3022 Traditional IP Network Address Translator
• RFC3489 STUN-Simple Traversal of User Datagram Protocol (UDP) Through Network Address Translators (NATs)
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