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 HST- 3000 s. One unit, set to AutoAnswer mode, is connected to the router at the customer premises. The second HST- 3000 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 Analog phone Legacy PABX Switch Figure 1. Testing with HST-3000 in a VoIP network. 2
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. 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. 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. 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 2. VoIP call set-up/signaling message summary Figure 3. Quality of service analysis 3
General specifications Test Ports/Interface Support 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 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 Gatekeeper Settings User Selectable Static/ Auto Discovery, or no gatekeeper direct connect mode Supports inbound and outbound calls with or without gatekeeper support. 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 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. Power supply 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 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) Figure 4. VoIP delay analysis Figure 5. Patented Telchemysingle ended voice call quality assessment 4
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 Acterna is present in more than 80 countries. To find your local sales office go to: www.acterna.com North America One Milestone Center Court Germantown, Maryland 20876-7100 USA Toll Free: 1 866 ACTERNA Toll Free: 1 866 228 3762 Tel: +13013531560x 2850 Fax: +1301353 9216 Latin America Acterna do Brasil Ltda. Av. Eng. Luis Carlos Berrini 936 9th Floor 04571-000 São Paulo SP-Brazil Tel: +55 11 5503 3800 Fax:+55 11 5505 1598 Asia Pacific Acterna Hong Kong Ltd. Room 902, 9th Floor Bank of East Asia Harbour View Centre 56 Gloucester Road Wanchai, Hong Kong Tel: +852 2892 0990 Fax:+852 2892 0770 Western Europe Acterna Germany GmbH 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 Acterna Moscow Prospect Mira 26, stroenie 5 RF-129090 Moscow Tel.: +7 095 937 88 04 Fax: +7 095 775 26 05 Copyright 2004 Acterna, LLC. All rights reserved. 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. Note: Specifications, terms and conditions are subject to change without notice. HSTVOIP/DS/ACC/08-04/AE/PDFONLY