Saturn 34 Series

Carrier-class VoIP Gateway

The brand new Saturn 34 Series is designed to carry both voice and facsimile over the IP network. It uses the industry standard SIP call control protocol so as to be compatible with free registration services or VoIP service providers’ systems. As a standard user agent, it is compatible with all common Soft Switches and SIP proxy servers. While running optional server software, the Saturn 34 Series can be configured to establish a private VoIP network over the Internet without a third-party SIP Proxy Server.

The Saturn 34 Series VoIP Gateway empowers service providers delivering carrier- class IP Centrex service over a Total-IP NGN / 3GPP IMS infrastructure in a more secured way.

The Saturn gateway provides a connection platform between traditional POTS lines and the Internet. Over various broadband technologies including xDSL, HFC, wireless and fiber, the Saturn 34 Series carries toll quality voice, fax and data traffic simultaneously in a cost effective way.

RFC 3261 SIP protocol VoIP Gateway

16 / 32 FXS or 16 / 32 FXO Lines

1WAN + 1LAN, RJ-45 10/100 baseT Ethernet

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, 3-Way Conference with/without media server, call parking and more

WEB based configuration (HTTP/HTTPs)

TR-069, TR-104, DHCP Auto Provision


IPv4, IPv6 support*


-   Saturn 3416S: 16 FXS, 2 Ethernet ports (1 WAN + 1 LAN)

-   Saturn 3416O: 16 FXO, 2 Ethernet ports (1 WAN + 1 LAN)

-   Saturn 3432S: 32 FXS, 2 Ethernet ports (1 WAN + 1 LAN)

-   Saturn 3432O: 32 FXO, 2 Ethernet ports (1 WAN + 1 LAN)

Voice Feature

-     G.722,G.711 a /μ-law, G.723.1, G.726, G.729A/B or GSM 6.10 Full Rate, ILBC13.3kbps

-     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)

-     Programmable Gain Control

-     Local Mixer

-     ITU-T V.152 Voice-band Data over IP Networks

SIP Method Support


SIP Call Features

-     Peer to Peer Call

-     Call Hold / Retrieve

-     Call Waiting

-     Call Pick Up

-     Call Park / Retrieve (with SIP Server)

-     Call Forward – unconditional, busy, no answer

-     Call Transfer – attended, unattended

-     Speed Dialing

-     Repeat Dialing

-     Three-way Calling

-     MWI (RFC-3842)

-     Hot Line and Warm Line

SIP Call Management

-     Support Outbound Proxy

-     Support up to three SIP servers

-     SIP Registration Automatic Failover

-     Group Hunting

-     Privacy Mechanism /Private Extension to SIP

-     Session Timers(Update / Re-invite)

-     DNS SRV Support

-     Call Types: Voice/ Modem/ FAX

-     Call Routing by Prefix Number

-     User Programmable Dial Plan

-     By routing Calling Number Manipulation

-     RADIUS CDR Client

-     Manual Peer Table(for P2P calls)

-     E.164 Numbering, ENUM support

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

Physical Interface

-     WAN: 1 x 10/100 Ethernet, auto cross-over, auto speed negotiation, RJ-45

-     LAN: 1 x 10/100 Ethernet, auto cross-over, auto speed negotiation, RJ-45

-     1 x RJ21 female connector for FXS line wiring

-     Power jack, Power switch, Reset button

Telephony Specification

-     In-Band DTMF, Out-of-Band DTMF Relay (RFC2833 or SIP INFO)

-     DTMF / PULSE Dial Support

-     Caller ID Generation(FXS) and Detection(FXO):

Ÿ     .DTMF

Ÿ     .FSK-Bellcore Type 1 & 2

Ÿ     .FSK-ETSI Type 1 & 2


Ÿ     .FSK: Calling Name, Number, Date and Time , VMWI

-     FXS metering pulse:

ŸŸ     .Polarity Reversal

ŸŸ     .12kHZ calling tone

ŸŸ     .16kHZ calling tone

-     Polarity Reverse

-     T.30 FAX bypass, T.38 Real-Time FAX Relay

-     FXS Line test and diagnostics with visual alarm indication

  Inward self test:

ŸŸ     .Loopback - codec

ŸŸ     .Loopback - analogue

ŸŸ     .Ringer

ŸŸ     .Outward Test (GR909 Standard):

ŸŸ     .REN

ŸŸ     .Phone Line disconnected

ŸŸ     .H.F. AC Voltage (Hazardous and foreign AC Voltage)

-     Fax and Modem over IP (up to 14,400bps)

-     Failsafe mechanism: FXS auto or manual relay to FXO through hardware relay or internal PCM Bus while Network, Service or power failure occurs

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

-     Loop Current Suppression

LED Indicators

-     Power, VoIP, Alarm, Run, WAN, LAN, Phone off-hook 1~32 / Phone Ch Alarm 1~32, Line 1~32


-     RJ21 to RJ11-male cable

-     RJ45 cable

-     AC Power cord

-     User Manual in CD

-     Rack mount kit

General Information

-     Dimensions : 44.5 (W) x 33 (D) x 4.5 cm (H)

-     Weight : 4.4 Kg

-     Power : AC 100~240 V 50/60Hz input

-     Operating temperature :0°C~ 45 °C

-     Storage temperature : - 25°C~ 75 °C

-     Operating Humidity : Up to 90% RH, non-condensing 


IP Network Specification

-     Support IPv4, IPv6 future upgradeable

-     WAN: Static IP, PPPoE, DHCP, PPtP

-     Network Protocol Support:


       HTTPS, DNS, DNS SRV, TelnetDHCP Server, DHCP Client, SNTU ClientUPnP,

       IGMP, IGMP snooping, IGMP proxy

QoS Support

-     WAN: DiffServ, IP Precedence

ŸŸ     .Priority Queue

ŸŸ     .Rate Control

ŸŸ     .802.1Q (VLAN Tagging), 802.1p (Priority Tag)

-     LAN: Rate Limit

DDNS Support

- (Dynamic and Custom)

-     TZO

-     Peanut Hull

Network Security Specifications

-     PPtP Client

-     DIGEST Authentication

-     MD5 Encryption

-     DoS Protection


-     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, TR104 (optical) 


SIP, Voice and FAX Related Standard

-     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

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

-     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)

-     RFC3261 SIP: Session Initiation Protocol

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

-     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

-     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 UPDATE)

-     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)