Voice Test Platform (VTP)
The Voice Test Platform (VTP) tests the continuity, and voice quality of Circuit-Switched (CS) and Packet-Switched (PS) voice calls. The platform verifies the speech path and, in each direction, is capable of transmitting reference sampled analog voice files from disk and recording their corresponding versions on disk after passing through the CS or PS networks. Multiple simultaneous calls are supported. The VTP includes a licensed Perceptual Objective Listening Quality Assessment (POLQA) algorithm to analyze the voice quality of recorded audio.
The VTP hardware consists of a host processor and an analog Foreign Exchange Office (FXO) telephony card seated in a 19” rack mountable chassis. The host processor is a commercial x86 PC, running the Microsoft Windows 7 (64-bit) operating system.
The Python-based script engine (CPython 3.5.3) in the VTP allows for automated, application-specific testing, carried out with test scripts. In addition to controlling the internal voice servers, the script engine can interface to other network elements such as a User Terminal, Network Emulator, or Physical Layer Tester to test end-to-end call scenarios. Users can quickly build test cases, using the familiar Python script language with a rich library of call processing APIs included with the platform. The analog and VoIP channels are highly configurable, allowing fine control over voice call test scenarios. The VoIP channels are supported by a full featured SIP stack, providing SIP Agent server and client features and a variety of codecs including G.729 and Opus. The FXO channels operate with North American and ITU Standards and variants covering most signalling and supervision features.
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FXO INTERFACE |
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Channels |
0 – 16 |
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Connector |
RJ21 telco (RJ11 via breakout box) |
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Compression |
µ-law A-law |
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Supervision |
Ring detection, Loop disconnect Reversal detection, Loop voltage, Loop current |
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Signaling |
Off hook, Flash, DTMF, Pulse dial |
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Protocols |
Bell 202 FSK Type 1 Protocol ITU-T V.23 FSK – British Telecom standard ITU-T V.23 FSK – General ETSI standard DTMF – General ETSI standard DTMF – Sweden/Finland variant DTMF – Denmark variant |
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Onhook Audio Detect |
Caller ID, DTMF, Audio logging |
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4-wire support |
Via external 2-4 wire converters |
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VoIP INTERFACE |
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Channels |
0 – 256 |
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SIP Compliance |
RFC 3261, RFC 3262, RFC 4028, RFC 3960, RFC 2976, RFC 2833, RFC 2782, RFC 3551, RFC 2474, RFC 5246 |
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Codecs |
ITU-T G.711 µ-law ITU-T G.711 A-law ITU-T G.729 GSM Adaptive Multi-Rate (AMR) GSM Enhanced Full-Rate (EFR) GSM Adaptive Multi-Rate Wideband (AMR-WB) Opus |
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Features |
Inbound/Outbound Registration Authentication NAT transversal Options method SDP parsing Subscribe/notify messages Customized SIP headers Configurable SIP timers Redirect responses |
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GENERAL PURPOSE INTERFACE |
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Ethernet |
1 x 10/100/1000 Base T |
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USB |
4 x USB 2.0, 2 x USB 3.0 |
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Video |
VGA, DVI |
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Audio |
Standard PC Audio |
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CAPABILITIES |
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Test control |
Python scripts |
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Script APIs |
Call processor, UT monitor & control, BGAN network emulator, POLQA |
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Audio paths |
Analog, file |
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Voice path continuity check |
Via tones or POLQA MOS, |
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Voice quality evaluation |
POLQA |
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MECHANICAL/ENVIRONMENTAL |
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Form factor |
19” / 2U rack mount |
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Size |
L 51 cm x W 43 cm x H 9 cm L 20 in x W 17 in x H 3.5 in |
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Weight |
Approx. 11 kg (24 lb) |
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Power connector |
IEC 320 male |
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Voltage |
100-240 VAC, 50/60 Hz |
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Current (typical) |
Approx. 0.6 A rms at 115 VAC |
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Operating temperature |
10°C to 35°C |
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Operating humidity |
20% to 90% relative humidity, non-condensing |
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Regulatory |
FCC, CE and RoHS compliant Safety: EN60950-1 Emissions: EN55022 Class A Immunity: EN55024 |
4017 Carling Avenue, Suite 200
Ottawa, ON, Canada K2K 2A3
Tel + 1 613 271 0044
Fax+ 1 613 271 3007