Testers & Simulators

Voice Test Platform (VTP)

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.

FXO INTERFACE

Channels

0 – 16

Connector

RJ21 telco (RJ11 via breakout box)

Compression

µ-law

A-law

Supervision

Ring detection, Loop disconnect

Reversal detection, Loop voltage,

Loop current

Signaling

Off hook, Flash, DTMF, Pulse dial

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

Onhook Audio Detect

Caller ID, DTMF, Audio logging

4-wire support

Via external 2-4 wire converters

VoIP INTERFACE

Channels

0 – 256

SIP Compliance

RFC 3261, RFC 3262, RFC 4028, RFC 3960, RFC 2976, RFC 2833, RFC 2782, RFC 3551, RFC 2474, RFC 5246

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

Features

Inbound/Outbound Registration

Authentication

NAT transversal

Options method

SDP parsing

Subscribe/notify messages

Customized SIP headers

Configurable SIP timers

Redirect responses

GENERAL PURPOSE INTERFACE

Ethernet

1 x 10/100/1000 Base T

USB

4 x USB 2.0, 2 x USB 3.0

Video

VGA, DVI

Audio

Standard PC Audio

CAPABILITIES

Test control

Python scripts

Script APIs

Call processor, UT monitor & control, BGAN network emulator, POLQA

Audio paths

Analog, file

Voice path continuity check

Via tones or POLQA MOS,

Voice quality evaluation

POLQA

MECHANICAL/ENVIRONMENTAL

Form factor

19” / 2U rack mount

Size

L 51 cm x W 43 cm x H 9 cm

L 20 in x W 17 in x H 3.5 in

Weight

Approx. 11 kg (24 lb)

Power connector

IEC 320 male

Voltage

100-240 VAC, 50/60 Hz

Current (typical)

Approx. 0.6 A rms at 115 VAC

Operating temperature

10°C to 35°C 

Operating humidity

20% to 90% relative humidity, non-condensing

Regulatory

FCC, CE and RoHS compliant

Safety: EN60950-1

Emissions: EN55022 Class A

Immunity: EN55024

Square Peg Communications Inc.
4017 Carling Avenue, Suite 200 
Ottawa, ON, Canada K2K 2A3
Tel + 1 613 271 0044
Fax+ 1 613 271 3007
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