The document details the configuration and features of the Matrix Comsec V1R1, focusing on the Setu ATA211 and ATA211G devices with capabilities such as SIP protocol support, dynamic jitter buffer, and various audio codecs. It outlines call handling, networking features like VLAN tagging and NAT support, as well as security measures including user authentication. Additionally, it mentions web-based administration and automated provisioning for maintenance ease.

1. Matrix ComSec V1R1
Matrix Linksys
Feature SETU ATA211 SETU ATA211G SPA3102
Configuration
FXS 1 1 1
FXO/GSM 1 FXO 1 GSM 1 FXO
Ethernet 2 2 2
SIP
Protocol Session Initiation Protocol (SIP) v2 Session Initiation Protocol (SIP) v2
SIP Proxy Redundancy Yes Yes
GSM
Frequency Bands Quad‐Band Support (SETU ATA211G)
Security Standards SIM PIN, IMEI Numbers (SETU ATA211G) No
C ll H dli
Call Handling RCOC R i ll i i l ll (SETU ATA211G)
RCOC: Routing call to original callee (SETU ATA211G)
Voice
G.729A, G.729B, G.723.1 (6.3 kbps, 5.3 kbps), G.711 (A‐law and μ‐ G.729 A, G.723.1 (6.3 kbps, 5.3 kbps), G.726 (16/24/32/40
Codecs
law) kbps), G.711 (A‐law and μ‐law)
If the G.729a codec is enabled and included in the codec
list, that resource is tied up until the end of the call
whether or not the call actually uses G.729a. If the G.729a
Simultaneous Calls simultaneous call ratio is not effected by codec selection resource is already allocated and since only one G.729a
resource is allowed per device, no other low‐bit‐rate
codec may be allocated for subsequent calls; the only
choices are G711a and G711u.
Dynamic Jitter Buffer (Adaptive), Full‐duplex audio, Speech Volume Dynamic Jitter Buffer (Adaptive), Comfort Noise
Settings, Comfort Noise Generation, Echo cancellation (G.168 with Generation, Echo cancellation (G.168 with 8/16/32ms Tail
Voice Quality
8/16/32ms Tail Length), Voice activity detection (VAD) with silence Length), Voice activity detection (VAD) with silence
suppression, Forward Error Correction suppression, Forward Error Correction
Arrange CODECS in preference order , also aids in effective
Vocoder Preference Limited Application
bandwidth utilization
Networking
IPv4 Yes Yes
DHCP Yes Yes
Point‐to‐Point Protocol over
Yes Yes
Ethernet (PPoE) client
VLAN tagging Yes Yes
NAT and STUN Support Yes Yes
Call Handling
Incoming Call Routing Any‐Any port Routing Incoming calls can be received on fixed port only
Automated Dial Plan helps Routing calls via specific port as per
Outgoing call Routing Calls can be routed via fixed SIP accounts only
dialed number
A dialplan with 100 entries, can be enabled for all ports or
Least Cost Routing Not Applicable
selectively
4 list, each of 8 entries can be programmed, ANT table can be
Automatic Number Translation 8, Fixed port wise
appllied port‐wise
Peer‐to‐Peer Call Table with 500 Entries facilating dialing
Peer‐to‐Peer Calls Abbreviated codes (instead of IP address) while making a peer‐to‐ No Such Provision
peer call
Proxy Calling 3 SIP Accounts (Service Line) 2 SIP Accounts (Service Line)
Call Progess Tone Generation
Country‐wise Defaults Provisioning through GUI Programmable Values
(CPTG)
Supplementary Services Yes Yes
Speed Dialing 20 Entries per port 8 Numbers
FAX
T.38 and PSTN pass‐through Yes Yes
Provisioning, Adminstration and Maintainance
Web browser administration and
configuration via integrated web Yes Yes
server
Telephone keypad configuration Yes Yes
Automated provisioning and
upgrade via HTTP, Trivial File Yes Yes
Transfer Protocol (TFTP)
Syslog Client, Multi Level
Syslog Client Multi‐Level Yes Yes
PCAP Trace‐Ethereal Capture to
know signalling with ITSP or other Yes Not Provided
system, without need of Hub
Security
Password Protection Yes Yes
Authenticate users before allowing internetwork calls, 100 User
PIN Authentication Only 8 users Accounts
Accounts
Digest Authentication‐Web standard
Yes Yes
for Authentication
QoS
Diffserv and ToS Yes Yes
The comparison is made as per the respective product collaterals available . For more details on a product please contact the manufacturer.