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Primus VoIP Network Extension
Imagine your network reaching out to your customers
anywhere in the world without the boundaries of time, money or distance.
Imagine having the ability to secure 100% of your global customer’s
business no matter where their locations are.
Stop imagining, the time is now. Primus VoIP Network
Extension allows your network to be anywhere and gives you the ability
to service 100% of your customer locations. No longer is your global
customer someone else’s customer.
Primus VoIP Network Extension leverages the Primus
Global VoIP and TDM networks-offering premium cost- effective service
to your customers. This SIP based service can be implemented in
hours and is proactively monitored and managed by our 24x7 global
service management center.
Primus VoIP Network Extension is the gateway to
the future of IP voice telephony
Product features:
- Secure, global VoIP and TDM network
- Primus Proprietary Centralized VoIP Solution
- Streamlined provisioning system
- Web-based management tool
- 24 x 7 x 365 Network Management Center
- Interoperability with multiple protocols and
architectures
Benefits to the carrier:
- Unrealized revenue potential within quick reach.
- Remove all geographic boundaries by providing
100% service to all local and global customers
- Manage own pricing that is attractive in local
and global markets
- Eliminate the need for costly origination alternatives
like international privae lines
- No additional network investment required
- Invoice end users with detailed call records
- Pure, private branding with end to end ownership
of the customer
- Immediately impact revenue via a streamlined
enrollment process that takes hours, not weeks to implement
- Reliable, quality voice service based on Primus’
VoIP and global TDM network.
- 24x7x365 network monitoring, troubleshooting
and customer support
Benefits to the carrier’s customer:
- Significant savings on globally originated
voice service on both inter and intra country calling
- Reduced operating expenses, overhead and maintenance
costs by using a common IP network to provide multiple services
- Worldwide implementation availability - virtually
anywhere Internet service is accessible
- Begin saving immediately with our streamlined
enrollment process that takes hours, not weeks to implement
- Reliable, quality voice service based on one
VoIP and global TDM network
- 24x7x365 network monitoring, troubleshooting
and customer support.
- Conduct business with a trusted carrier anywhere
in the world and simplify their global telecom needs.
Primus Small Business Solution
The Primus Small business solution typically
is designed for small satellite offices with
one or two lines that use the Cisco ATA 186. The Cisco ATA’s
are stackable to
accommodate more lines without having to upgrade to the enterprise
solution, which
requires equipment that will interface with a PBX.
Primus Enterprise Solution
The Primus Enterprise solution is designed for
medium and larger sized offices with a PBX. This solution can be
scaled from 8 lines up to a full T1/E1 configuration. The Vega 50
can accommodate any traffic routed to it off the PBX. This allows
for seamless, centralized routing for an entire office.
Primus VoIP Network Extension
The Primus VoIP Network uses SIP protocol to set up,
authenticate, and complete calls over our IP network. The network
consists of a SIP server which routes calls, collects CDRs and provides
a feed to our billing and web based monitoring
systems. The calls are routed to ensure the best quality and often
times they are taken off our IP network and terminated over the
Primus TDM network. The routing intelligence of the network allows
Primus to route the call in the most direct and
efficient way. The Primus network allows for an IP originated call
to terminate to any PSTN phone in the world. Primus has a vast VoIP
and TDM network that allows for redundant coverage to most destinations.
Primus VoIP Network Extension - Equipment Options
Small
Business Solution
Cisco ATA 186
Voice-over-IP
(VoIP) protocols
- H.323
v2
- H.323
v4
- SIP
(RFC 2543 bis)
- MGCP
1.0 (RFC 2705)
- MGCP
1.0/network-based call signaling (NCS) 1.0 Profile
- MGCP
0.1
- SCCP
Voice
coder-decoders (codecs)
- G.729,
G.729A, G.729AB1
- G.723.1
- G.711A
- G.711µ
Provisioning
and configuration
- DHCP
(RFC 2131)
- Web
configuration via built-in Web server
- Touch-tone
telephone keypad configuration with voice prompt
- Basic
boot provisioning (RFC 1350 TFTP Profiling)
- Dial
plan provisioning
- Cisco
Discovery Protocol for SCCP
Security
- H.235
for H.323
- RC4
encryption for TFTP configuration profiles
Dual-tone
multi-frequency (DTMF)
- DTMF
tone detection and generation
- Out-of-band
DTMF H.245
- Out-of-band
DTMF for H.323
- RFC
2833 AVT tones for SIP, MGCP, SCCP
Call
progress tones
- Configurable
for two sets of frequencies and single set of on/off cadence
Line-echo
cancellation
- Echo
canceller for each port
- 8
ms echo length
- Nonlinear
echo suppression (ERL greater than 28 dB for f = 300 to
3400 Hz)
- Convergence
time = 250 ms
- ERLE
= 10 to 20 dB
- Double-talk
detection
Voice
features
- Voice
activity detection (VAD)
- Comfort
noise generation (CNG)
- Dynamic
jitter buffer (adaptive)
Fax
- G.711
fax pass-through2
- G.711
fax mode2
Dimensions
- 1.5
x 6.5 x 5.75 in. (3.8 x 16.5 x14.6 cm) (H x W x D)
Weight
Power
- Power
consumption
- 0.25
to 7.5 W (idle to peak)
- DC
input voltage
- +5.0
VDC at 1.5 A maximum
- Power
adaptor
- Universal
AC/DC
- ~
3.3 x 2.0 x 1.3 in. (~8.5 x 5.0 x 3.2 cm)
- ~
4.8 oz (135 gm) for the AC-input external power adaptor
- ~
4 ft (1.2 m) DC cord
- ~
6 ft (1.8 m) cord
- UL/CUL,
CE approved
- Class
II transformer
Physical
interfaces
- Ethernet
- RJ-45
8-wire connector, IEEE 802.3 10-Base-T standard Analog
telephone
- Two
RJ-11 FXS voice ports
- Power
Indicators
- Function
button with integrated status indicator
- Activity
LED indicating network activity
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Enterprise
Solution
Vegastream – Vega 50 BRI ISDN/ Analog
Voice-over-IP
(VoIP) protocols
Voice
coder-decoders (codecs)
- G.723.1
5.3/6.4
- G.729a
8
- G.711
(a-law/µ-law) 64
- Auto
Selecting
- Voice
Activity Detection
- Comfort
Noise Generation
- Silence
Suppression
LAN
Interface
- 10
BaseT, 100 BaseTX, full or half duplex
Telephony
Interface
- ETSI
BRI
- 4
x S/T interfaces
- 8
simultaneous calls
- Point
to point
- Point
to Multipoint
- Each
Interface can be NT or TE
Telephony
Features
- Caller
ID
- Caller
ID Screening guarantees
connection only from authenticated call sources
- Dial
Planner
- Multiple
Subscriber Number (MSN)
- Direct
Dialling In (DDI)
Typical
Unit Latency
- 135ms
for G.723.1
- 55ms
for G.729a
- 50ms
for G.711
Fax Support
Modem
Support
Echo
Cancellation
- G.165/G.168
up to 32ms of echo cancellation
Dial
Planner
- Sophisticated
call routing capabilities
- Standalone
or Gatekeeper operation
Physical
Dimensions
- 440mm
(17.4") x 86mm (3.4") x 303mm
- (12")
width/height/depth.
- Industrial
rackmount: 483mm (19")
- Weight:
7kg
Processors
- DSP
– TMS 320VC549 / TMS320VC5409
- CPU
– IDT 79RV4640, MIPS RISC
Operating
System
- Nucleus
embedded system for RISC
Processors
Power
- 100-240
VAC.47-63 Hz.1A-0.5A
Front
Panel Display
- LED
– ISDN: DSL Physicals
– LAN: Speed, Activity, Half/Full Duplex
- LCD
– IP address, host name, Call Activity
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Product Information
Collateral
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