Description
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- RFC 3261 SIP protocol VoIP Gateway
- 8 FXS / 8PSTN and 5 Ethernet ports
- 1WAN + 4LAN, RJ-45 10/100Ethernet
- Interactive Voice Response (IVR)
- T.30 and T.38 compliant
- Line reversal and metering tone (12K/16KHz)
- Ethernet switch function with QoS and VLAN
- IGMP Proxy/Snooping
- Advanced calling features, call parking and more
- WEB based configuration (HTTP/HTTPs)
- TR-069, TR-104, DHCP Auto Provision
- SNMP V2c/V1
- Busy Tone cadence auto learning/detection
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Model
- 8 FXS, 5 Ethernet ports (1 WAN + 4 LAN)
- 8 PSTN, 5 Ethernet ports (1 WAN + 4 LAN)
Voice Feature
- G.722, G.711 a/μ-law, G.729A/B, G.726, G.723.1, GSM 6.10 Full Rate, iLBC 13.3 kbps
- DTMF Detection and Generation
- Silence Suppression & Detection
- Comfort Noise Generation (CNG)
- Voice Activity Detection (VAD)
- Echo Cancellation (G.165/G.168)
- Adaptive (Dynamic) Jitter Buffer
- Call progress tone generation (FXS)
- Auto or Programmable Gain Control
- Inbuilt Local Mixer
- ITU-T V.152 Voice-band Data over IP Networks
SIP Method Support
- ACK, BYE, CANCEL, INFO, INVITE, MESSAGE, NOTIFY, OPTIONS, PING, PRACK, PUBLISH, REFER, REGISTER, SUBSCRIBE, UPDATE
SIP Call Features
- Peer to Peer Call
- Call Hold / Retrieve
- Call Waiting
- Call Pick Up
- Call Park / Retrieve (SIP Server Required)
- Call Forward – unconditional, busy, no answer
- Call Transfer – attended, unattended
- Do Not Disturb
- Speed Dialing
- Repeat Dialing
- Three-way Calling
- MWI (RFC-3842)
- Hot Line and Warm Line
Telephony Specifications
- In-Band DTMF, Out-of-Band DTMF Relay (RFC2833 or SIP INFO)
- DTMF / PULSE Dial Support
- Caller ID Generation / Detection:
DTMF
FSK-Bellcore Type 1 & 2
FSK-ETSI Type 1 & 2
FSK-NTT
FSK: Calling Name, Number, Date and Time, VMWI- FXS metering pulse:
Polarity Reversal
12kHz calling tone
16kHz calling tone- Polarity Reversal Detection (FXO) and Generation (FXS)
- T.30 FAX Bypass, T.38 Real Time FAX Relay
- FXS Line test and diagnostics with visual alarm indication
※Inward self test:
Loopback – codec
Loopback – analogue
SLIC DC power voltage
Tip / Ring DC feed
Ringer
※Outward Test (GR909 Standard):
REN
Phone Line disconnected
H.F. DC Voltage (Hazardous and foreign DC Voltage)
H.F. AC Voltage (Hazardous and foreign AC Voltage)
Tip / Ring Short- Failsafe mechanism: FXS auto or manual relay to FXO /PSTN through hardware relay or internal PCM Bus while Network, Service or power failure occurs
- Emergency Number Table (PSTN)
- Modem over IP up to 14,400bps
- ROH Tone (Receiver Off-Hook Tone @ 480 Hz)
- Loop Current Suppression
SIP Account Management
- By port registration
- By device registration (share account)
- Mixed mode (Hunt number for inbound, by port number for outbound)
- Invite with Challenge
- Register by SIP Server IP Address or Domain Name
- Support RFC3986 SIP URI format
SIP Call Management
- Support Outbound Proxy
- Register up to three SIP servers
- SIP Registration Failover Mechanism
- Group Hunting
- Privacy Mechanism / Private Extensions to SIP
- Session Timers (Update / Re-invite)
- DNS SRV Support
- Call Types: Voice / Modem / FAX
- Call Routing by Prefix Number
- User Programmable Dial Plan Support
- Toll-Free Support (PSTN)
- Automatic Calling Number Manipulation (VoIP & PSTN)
- CDR Client
- Manual Peer Table (for P2P calls)
- E.164 Numbering, ENUM support
Physical Interface >>
- WAN : 1 x 10/100/1000M Ethernet interface, auto cross-over, auto speed negotiation, RJ-45 connector
- LAN : 4 x 10/100/1000M Ethernet interface, auto cross-over, auto speed negotiation, RJ-45 connector
- 8 RJ11 connectors for FXS/PSTN line wiring
- Power jack, power switch
- Reset button
LED Indicators
- Power, Provision/Alarm, Register, WAN, LAN1~LAN4, Phone off-hook 1~8 / Phone Ch Alarm 1~8, Line1~8
IP Network Specifications
- WAN: Static IP, PPPoE, DHCP, PPTP
- Network Protocol Support:
IP, TCP, UDP, TFTP, FTP, RTP, RTCP, RTCP XR, ARP,RARP, ICMP, NTP, SNTP, HTTP,
HTTPS, DNS, DNS SRV, Telnet, DHCP Server, DHCP Client, STUN Client, UPnP,
IGMP, IGMP snooping, IGMP proxy, RTSP ALG, SIP ALG- NAT Functions
Support up to 255 Clients
Port Forwarding (Virtual Servers)
DMZ
Port Triggering- Support IPv4, IPv6 future upgradeable
- QoS Support:
※WAN: DiffServ, IP Precedence
Priority Queue
Rate Control
※LAN: Rate Limit- DDNS Support
Dyndns.org (Dynamic and Custom)
Network Security Specifications
- PPTP Client
- DIGEST Authentication
- MD5 Encryption
- DoS Protection
Management
- Web Based Configuration
- Auto-provisioning (HTTP / HTTPS / TFTP)
- Telnet
- IVR
- FTP / TFTP / HTTP Software Upgrade
- Configuration Backup and Restore
- Reset to Default Button
- TR-069/104 (Option)
- SNMP V3/ V2c/ V1
SIP, Voice and FAX Related Standard
- 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 Session Initiation 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 Related Standard
- 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)
- RFC2782 A DNS RR for Specifying the location of Services (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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