Acterna HST-3000

Transcription

Acterna HST-3000
Acterna HST-3000
Option for VoIP
Voice over Internet Protocol (VoIP)
services are now being widely
deployed. Quick, efficient and cost
effective installation of these services
is therefore a key requirement. A
simple standards based field tool
enabling technicians to verify the
service, check voice quality and
troubleshoot problems is therefore an
essential tool to support VoIP service
deployment.
The HST-3000 VoIP tester is a versatile
field solution for VoIP service turn-up
and troubleshooting. Handheld,
rugged and easy-to-use, the HST-3000
can validate VoIP service connectivity,
feature availability and voice quality.
In addition, it provides comprehensive
features, including signaling, IP ping,
packet statistic and trace route
analysis to identify, diagnose and
sectionalize VoIP network and
equipment problems.
Advanced automated test functions,
custom scripting and one-button
operation also ensure consistent,
accurate and repeatable test methods
are used. This allows for rapid,
efficient and cost-effective delivery
of VoIP.
Highlights
– VoIP phone emulation for service
turn-up and troubleshooting
– VoIP voice quality assessment and
rating using the patented Telchemy
single-ended live call quality
assessment method
– Supports Cisco to SCCP, SIP and
H.323 signaling protocols
– Real-time VoIP and Ethernet
statistics
– IP Ping and ICMP/UAP trace route
testing
– AutoAnswer mode for two-ended
testing that requires only one
technician
– Modular hardware and software
architecture that is flexible and
easily upgraded to allow testing of
multiple services
– CE-marked
VoIP Service Turn-up
To ensure successful VoIP service
turn-up, connectivity to the signaling
gateway, feature availability and call
quality must be proven.
The simplest and fastest way to verify
connectivity is to place an actual VoIP
call. The HST-3000 can emulate an IP
phone and supports placing and
receiving VoIP calls utilizing Cisco to
SCCP, SIP and H.323 signaling gateways. Calls can be placed to different
endpoints to verify translation
provisioning - IP phone to IP phone,
site to site, IP to TDM network, to a
provisioned automated test line or to
another HST-3000, either manned or in
AutoAnswer mode.
Both subjective and objective voice
quality measurements can be gathered
during these connectivity test calls. The
HST-3000 provides a packet-based
objective measurement of VoIP call
quality by analyzing delay, jitter, and
packet loss to generate a good, fair or
poor quality rating based on configurable Quality of Service (QoS) score
thresholds.
Additionally, the HST-3000 uses the
patented Telchemy single-ended live
call method to provide a real-time
assessment of subjective voice quality
in terms of both Mean Opinion Score
(MOS) and R Factor. The valuable data
provided by this analysis is compared
with the data from the objective measurements. This comparison helps to
quickly verify acceptable VoIP call quality. Call feature provisioning and the
availability of supplementary services
can also be verified during these calls.
The HST-3000 can also be used to turnup a VoIP service when it does not have
the required gateway signaling support. This procedure requires
end-to-end testing using two HST3000’s. One unit, set to AutoAnswer
mode, is connected to the router at the
customer premises. The second HST3000 unit is used to place calls back to
the first unit from different endpoints.
The first unit answers the call and plays
a pre-recorded message. Test measurements are obtained from the
pre-recorded message and displayed
on the HST-3000.
Customer Premise
Exchange
IAD
Router
Soft switch
Soft phone
Router
IP phone
PSTN
Legacy PABX
Analog phone
Figure 1. Testing with HST-3000 in a VoIP network.
2
Switch
VoIP Troubleshooting
When a call is unsuccessful or the voice
quality is poor, the HST-3000 can be
used to identify and sectionalize problems.
With call-set-up problems it is essential
to have the ability to troubleshoot the
call signaling process. The HST-3000
provides real-time visibility of the
entire call set-up process. Signaling
message decodes and signaling error
messages are displayed allowing problems to be quickly and easily
pinpointed.
IP Ping and ICMP or UDP trace route
analysis can be performed to isolate
path/device connectivity problems
and sources of delay. Call throughput
can be measured. Additionally, Ethernet statistics are generated to aid the
diagnosis of call quality and identify
failed devices or network over utilization.
Figure 2. VoIP call set-up/signaling message summary
Reduce Costs, Increase
Productivity and Improve
Efficiency
The HST-3000 provides a number of
powerful features that can greatly
improve the VoIP service turn-up and
troubleshooting process, reducing
costs and improving productivity and
efficiency.
The HST-3000’s straightforward graphical user interface (GUI) greatly
simplifies the testing process, thus
reducing the amount of training needed for comprehensive testing.
The HST-3000’s pre-programmed tests
and customized scripts ensure that all
technicians follow the same procedures, speeding-up service delivery
and minimizing installation and testing
errors.
Standard Ethernet, USB and serial connections offer flexibility to easily
download software and offload stored
test results for later analysis.
In AutoAnswer mode two-ended testing across the VoIP network can be
accomplished by using a single technician. Additionally, one-button
automatic testing combined with support for all phases of VoIP service
deployment, reduces the number of
technicians required to turn-up and
troubleshoot service, as well as making it possible for non-experts to
operate tests.
Figure 3. Quality of service analysis
3
General specifications
Test Ports/Interface Support
Gatekeeper Settings
Power supply
10/100 Ethernet (configurable -Half/Full Duplex
auto detect), RJ-45
ADSL, G.SHDSL( Modem port 8 pin modular - line
on center pins) and T1
USB 1.1 Host
RS-232 9 pin DIN serial port
User Selectable Static/ Auto Discovery, or no gatekeeper direct connect mode
Supports inbound and outbound calls with or without gatekeeper support.
Batteries Lithium Ion, removable battery pack
VoIP Operating time approx. 6 to 8 hours of typical
usage
Auto switch-off 1 to 15 minutes after last action or
off
Charging time (internal) 7 hours from empty to full
charge
AC line operation via external adapter/charger
LCD Backlit Monochrome 320x240 display
Supported Signaling Protocols
H.323 ITU-T H.323 version 3 Fast Connect
H.323 ITU-T H.323 version 3 Full Connect
(MSD, CAPSET, OLC exchange)
RTP/RTCP RFC 1889 and 1890
Skinny Cisco Client Protocol(SCCP)
SIP RFC 3621
Supported Codec Configurations
ITU-T G.711 u-law/A-law(PCM/64kbt/s)
ITU-T G.723.1(ACELP/5.3,6.3 kbt/s)
ITU-T G.726(ADPCM/16,24,32,40kbt/s)
ITU-T G.729a(GS-ACELP/8kbt/s)
ITU-T G.729ab(GS-ACELP/8kbt/s
ITU- GSM – FR
ITU GSM - EFR
User selectable Silence Suppression, Jitter Buffer,
and voice packet size.
User selectable transmit source (Live Voice conversation, tone transmit(200-5kHz), pre-recorded wave
file(Up to 2Mbt)
LAN Settings
User-selectable Calling Alias
User-selectable IP address, static or DHCP
User-selectable subnet mask, gateway and DNS
server
User-selectable or default MAC address
VLAN configurable - IEEE.802.1p/q
Configurable IP TOS
Figure 4. VoIP delay analysis
4
Reported Results - VoIP
Full incoming call statistics, including IP address,
Alias, Name, RTCP availability/ports, Codec and
rate, call signaling support, silence suppression
enabled, and call duration
Throughput sent/received in bytes and packets, out
of sequence packets
Call progress and signaling error messages
Packet delay (min/max/avg)
Packet jitter (min/max/avg)
Packet loss (min/max/avg)
Encoding, packetization, buffering, and total delay
Voice Quality Rating based on packet metrics
thresholds set by user
MOS rating, R Factor, and Voice Degradation Factors
Permissible ambient temperature
Nominal range of use –14°C to +50°C
Storage and transport –25°C to +70°C
Dimensions (w x h x d)
240 x 114 x 70 mm
Weight including batteries
1.3 kg (3 lbs)
Reported Results - Ethernet TE
Link status, Link speed, link duplex detection.
Ethernet Statistics; collisions, TX/ RX( bytes,
frames, errors, dropped).
PING ICMP and UDP Statistics; Echos sent/received,
PING delay (cur/ave/max/min), lost count/percentage.
Supports IP address or DNS name destination.
Trace Route ICMP and UDP Statistics: Hop count,
name lookup, and IP address of hops.
Supports IP address and DNS address destination.
Figure 5. Patented Telchemy single ended voice call quality assessment
Ordering information
Base units
HST-3000C HST-3000C-CE base with copper testing
Requires the purchase of a SIM
HST-3000 HST-3000-CE base without copper testing
Requires the purchase of a SIM
SIMS (Modules)
HST3000-4WLL
4 wire local loop
HST3000-T1
Dual Tx/Rx bantam T1 interface and T1 software option
HST3000-CT1
Dual T/R/G interface for copper Testing and Dual Tx/Rx bantam T1
HST3000-T3
Dual Tx/Rx bantam T1 interface, and dual Rx, single Tx BNC DS3 interface
and DS3 software option
HST3000-BRI
ISDN BRI option
HST-ARCE
ADSL (ATU-R) option
HST-CAR
Copper (ATU-R) option
HST-CU
Dual T/R/G Interface to copper test option
HST-GSH
G.SHDSL option
HST3000-CuCE
Cu only SIM, CE mark
HST3000-CARCE
Cu & ATU-R (Annex A) SIM, CE mark
HST3000-ARCA
ATU-R/C dual mode SIM, AoPOTS
HST3000-CARCA
Cu & ATU-R/C dual mode SIM, AoPOTS
HST3000-ARB
Annex B ATU-R SIM
HST3000-CARB
Annex B Cu/ATU-R SIM
HST3000-ARCB
ATU-R/C dual mode SIM, AoISDN
HST3000-CARCB
Cu & ATU-R/C dual mode SIM, AoISDN
HST3000-CSHCE
G.SHDSL & Cu SIM
HST3000-BLK
Blank SIM
Software options
HST3000S-IP
Advanced IP suite – PING and through mode support
HST3000S-WEB
Web browser option
HST3000-WBTONES
WB TIMS option
HST3000-RFP
RFL option
HST3000S-VOIP
VoIP software
HST3000-FTP
FTP software option
HST3000-SCRIPT
Scripted test option
HST3000S-H.323
VoIP Signaling call contols for H.323
HST300S-SCCP
VoIP Signaling option for Cisco SCCP
HST3000S-SIP
VoIP Signaling option for SIP call control
HST3000-PCMSIG
Signaling (PCM) software option
HST3000-PCMTIMS
TIMS (PCM) software option
HST3000-T1DDS
DDS-T1 software option
HST3000-PRI
ISDN PRI software option
HST3000-SPE
Spectral Noise software option
HST3000S-MOS
MOS (Mean Opinion Score) Analysis option
HST3000-TDR
TDR Option
5
Worldwide
Headquarters
Regional Sales
Headquarters
One Milestone Center Court
Germantown, Maryland
20876-7100
USA
North America
One Milestone Center Court
Germantown, Maryland
20876-7100
USA
Toll Free: 1 866 ACTERNA
Toll Free: 1 866 228 3762
Tel: +1 301 353 1560 x 2850
Fax: +1 301 353 9216
Acterna is present in more
than 80 countries. To find
your local sales office go to:
www.acterna.com
Latin America
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936 9th Floor
04571-000 São Paulo
SP-Brazil
Tel: +55 11 5503 3800
Fax:+55 11 5505 1598
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Room 902, 9th Floor
Bank of East Asia
Harbour View Centre
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Tel: +852 2892 0990
Fax:+852 2892 0770
Western Europe
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Mühleweg 5
Germany
Tel: +49 7121 86 2222
Fax:+49 7121 86 1222
Eastern Europe,
Middle East & Africa
Acterna Austria GmbH
Aredstrasse 16-18
A-2544 Leobersdorf
Tel.: +43 2256 65610
Fax: +43 2256 65610-22
© Copyright 2004
Acterna, LLC.
All rights reserved.
Note: Specifications,
terms and conditions
are subject to change
without notice.
Acterna, Communications
Test and Management
Solutions, and its logo are
trademarks of Acterna,
LLC. All other trademarks
and registered trademarks
are the property of their
respective owners. Major
Acterna operations sites
are IS0 9001 registered.
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Prospect Mira 26,
stroenie 5
RF-129090 Moscow
Tel.: +7 095 937 88 04
Fax: +7 095 775 26 05
HSTVOIP/DS/ACC/08-04/AE/PDFONLY