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RTP Toolbox™
RTP Packet Testing & Simulation Tools



RTP ToolBox™ Software Ver 2.11.9 is Now Available! Download it here

Overview | Main Features | Create and Manage RTP Sessions | Generation/Detection of RTP Traffic
SIP Call Generation & Reception Capability | Analysis | Server - Client Functionality
Speech Quality of Packets | Applications | Other VoIP Products | Buyer's Guide


Overview

GL's RTP ToolBox™ testing and simulation tool is designed not only to monitor RTP and RTCP packets, but also to allow users to manually create and terminate RTP sessions, independent of call-signaling protocols such as SIP, H323, MEGACO, or MGCP.

This tool can be used for testing and developing enhanced voice features (VAD, echo cancellation, codec, digit regeneration, digit generation, fax over IP, jitter implementation etc) within end-user equipment (IP phones, ATA, MTA etc), testing media gateway telephony interfaces, end-to-end network testing before and during VoIP deployment, automated testing of digital signal processing embedded into network elements.

 
 
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RTP ToolBox™ Testing Applications


Main Features

  • Create RTP sessions & Auto scan incoming RTP sessions; supports IPv6 addressing.
  • Supported Codec's:
    • G.711 (µ-law/A-law - 64kbps)
    • G.711 App II (ALaw and µ-Law with VAD)
    • G.729, G.729B (8 kbps)
    • G.726 (5 bit 40 kbps/4 bit 32 kbps/3 bit 24 kbps/2 bit 16 kbps)
    • G.726 (40/32/24/16 kbps with VAD)
    • GSM (13.2 kbps)
    • AMR (4.75kbps, 5.15kbps, 5.9kbps, 6.7kbps, 7.4kbps, 7.95kbps, 10.2kbps, 12.2 kbps) (requires additional license)
    • AMR WB (Wideband codec - 6.60kbps, 8.85kbps, 12.65kbps, 14.25kbps, 15.85kbps, 18.25kbps, 19.85kbps, 23.05kbps, 23.85 kbps) (requires additional license)
    • EVRC (Rates - 1/8, ½ and 1), EVRC0 (requires additional license)
    • EVRCB (Rates - 1/8, ¼, ½ and 1), EVRCB0 (requires additional license)
    • EVRC_C (requires additional license)
    • SMV (Modes - 0, 1, 2 and 3- Available if licenses are provided or owned, please call GL)
    • iLBC (15.2kbps and 13.33kbps)
    • SPEEX (Narrow Band, Wideband))
    • G.722
    • G.722.1 (24 kbps and 32 kbps Wideband)
    • GSM HR (rate – 5.6kbps, packet time multiples of 20msec.)
    • GSM EFR (rate - 12.2kbps, packet time fixed at 20msec.)

    Click here for more comprehensive information on the above codecs.

  • Set the RTP traffic properties (payload type, codec) and impairments during Auto-scan of all incoming RTP traffic

  • Automatic scan option to capture all incoming RTP traffic.

  • Monitoring RTP streams using scalable Oscilloscope and Spectrum Analyzer.

  • Generation/Detection of in-band and out-of-band Digits/Tones (DTMF, MF, user-defined, etc.)/Events per RFC-2833 & RFC-4733 Set Delay and Attenuate for loop backed RTP traffic

  • User-defined impairments: latency, packet loss, out of sequence and duplicate packets.

  • Detailed statistical information of RTP and RTCP packets.

  • Sending and recording of voice files (.glw) with a synchronous TX/RX option.

  • G.168 testing for echo cancellation equipment.

  • Talk and Play to Speaker options using PC sound card.

  • Call Generation and Reception ability provides UA simulation (up to 8 UAs through CLI).

  • Customize Codec options (payload type, ptime) for UA during Call Generation (Dial) & Call Reception

  • Quality Metrics with R-Factor and MOS Factors, Jitter Buffer Statistics, Degradation Factor, and Burst Metrics are graphically represented

  • Supports Client-Server functionality (requires additional license)-C++, & TCL clients

  • Can run on any PC with Windows® XP (32 bit and 64 bit)/Vista (32 bit)/7 (32 bit and 64 bit) OS.

  • Automate the IVR testing process - call establishment and traffic generation / detection process through scripts

  • Monitoring IVR System for voice and data quality

Create and Manage RTP Sessions

The RTP ToolBox™ application provides following functionalities

  • Create an RTP Session

    An RTP session can be created either ‘Manually’ or using ‘Auto-Scan for Incoming Session’ feature. With Auto-scan feature, the application monitors all incoming packets addressed to the machine on which RTP ToolBox™ is running. If there are any RTP packets in the traffic, then the sessions on which these RTP packets are being transmitted are automatically displayed. For all the codecs, the payload type should match with the values set in the incoming RTP sessions at the transmission end.



    Click to Enlarge

  • Manage an RTP Session

    Set RTP Packet Properties

    It is possible to configure the properties for sending/receiving RTP Traffic with Tx/Rx profile option. On transmitting session, users can set the type of codec needed, sampling rate, voice payload type, RFC 2833 payload type, comfort noise payload type, packetization time, SSRC, timestamp, and sequence number for the out going Traffic. Users can also assign 'Quality Of Service (QoS)', i.e., IP Type of Service properties such as precedence, delay, throughput, and reliability values to the stream. On receiving session, users can specify a desired voice payload type for each codec, payload type to receive RFC 2833 events, Comfort Noise Payload Type, and set the buffer used for delayed packets that arrive at receiving end (Both static and dynamic jitter buffers are supported).



    Click to Enlarge

    Set Impairments

    Users can manually introduce Impairments and transmit on the RTP sessions. This includes introducing fixed latency, uniform/normal distributed latency, periodic/random/burst packet loss, out-of-order packets, and duplicate packets. Users may also apply delay and attenuate to the incoming data on a scanned session.

    Scripted Control

    Users can create a script that defines the RTP behavior. Scripting provides the users, a greater flexibility to combine Traffic actions with simple programmed scripts. On call establishment, this script can be loaded and executed as shown in the figure above. This option will also allow users to run the scripts automatically on the scanned sessions. Scripts can also be run on multiple sessions at the same time and its progress can be viewed in the Script Contents pane by highlighting the currently executing command of the script. For enhanced testing, users can also write IVR (Interactive Voice Response) scripts.


Generation/Detection of RTP Traffic

Generation/Detection of Digits/Tones/RTP Events

RTPToolBox™ application can be used to generate in-band digits and tones. The supported tones include single, dual, and multi-tones. Supported digits include DTMF, MF, and MFR2 forward and backward digits. The generation of RTP Events/Digits per RFC-2833 & RFC-4733 are also available.

Similar to generation, RTPToolBox™ application allows users to capture tones and digits in the traffic. It also displays additional information about the captured signal such as type of the signal, timestamp, event, power, and accept/ reject frequencies. This is completely supported for both in-band digits/tones and RTP digits/events per RFC-2833 & RFC-4733.



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Transmit/Record Voice File

The application can also record the incoming voice data to file. These files can be compared with GL's optional Voice Quality Testing software, providing PESQ, PAMS and PSQM score. The ability to send and record files also allows G.168 testing for echo cancellers.


SIP Call Generation & Reception Capability

RTPToolBox™ allows users to configure and simulate a user agent (UA) for manual SIP call generation and reception using public URL and contact IP addresses. Multiple SIP calls can be placed and received through a single user agent. All the calls at the application end will be answered automatically so received calls show the status as Hang-Up indicating that call has been established. The default RTP Port No., for the added call generation sessions is 9002. Up to 8 User Agents can be configured using the CLI



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The user can configure some codecs for call generation or reception through "CodecOptions.ini" file located in the installation folder. This file contains options such as packetisation time (ptime), payload type, and codec options. It also allows the user to select the codecs to be included in the test scenario, i.e, if the test requires only A-law or u-law be offered then those can be selected and the others omitted.

The codec packaging details can be edited in the CodecOptions.ini file directly, with the following restrictions

  • [CODEC_TYPE] - Allows user to select codecs to be offered in call setup.
  • [Packetisation_Time] - Minimum Packetisation time varies for codecs.
  • Payload Type – Unique value for each codec

Analysis

Oscilloscope and Spectrum Analyzer

In oscilloscope the PCM codes (amplitude of the incoming signal) for any selected session are graphically displayed in real-time as a function of time.

The spectrum analyzer displays data received in spectral domain (Spectral Amplitude vs. Frequency). A Fast Fourier Transform (FFT) is applied to successive sample sets of the incoming data and displayed in graphic form. The FFT length can adjust the frequency resolution. (from 32 points to 8192 points).



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Jitter Buffer Statistics, Quality Metrics (R Factor & MOS), Degradation Factor, Burst Metrics

Jitter Buffer feature allows you to set the buffer used for delayed packets that arrive at receiving end. Both static and dynamic jitter buffers are supported.

Quality metrics include various graphs for R-Factor and for MOS Factor. R Factor graph will display statistics such as, R-Listening, R-Conversational, R-G107 and R-Nom. MOS Factor graph will display statistics such as MOS CQ, MOS PQ and MOS Nom. The RTP Toolbox™ MOS call quality reporting is based on the Telchemy VQMon / SA application that reports call quality estimates including Listening and Conversational Quality MOS scores - MOS-LQ, MOS-CQ , and Listening and Conversational Quality R factors - R-LQ, R-CQ. Estimates are based on the ITU G.107 E Model

Degradation Factor statistics indicate Good quality packets, Packet loss, Packets discarded, Echo level and Regency. In addition to these statistics, RTPToolBox™ also supports Burst Metrics.



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RTP/RTCP Packet Statistics

Statistics reports of RTP and RTCP packets transmitted on a session such as Number of Packets Sent, Packets Received, Dropped Packets, Out of Sequence Packets, Sender Reports, and Receiver Reports are also displayed using RTP/RTCP statistics applications.


Server-Client Functionality (requires additional license)

RTPToolBox™ can be configured as server-side application, to enable remote controlling of the application through multiple command-line based clients. Supported clients include C++ and TCL based clients. User can remotely perform all functions such as creating RTP sessions, Digit/Tones/Event generation and reception, setting impairments, creating session profiles & so on. User can also generate and receive SIP calls through commands. The RTP sessions associated with the SIP call are created automatically.



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Sample Scripts - C++

  • Sample C++ Script 1 - illustrates how to monitor and transmit digits, tones, voice files, rfc2833 events and more.

  • Sample C++ Script 2 - illustrates the process of configuring user agents, establish a call and scanning for traffic.

Sample Scripts - TCL

  • Sample TCL Script - illustrates the user in creating two new sessions, where one session is for transmitting the inband digits and the other session is for monitoring the transmitted digits in the prior session

Estimating Speech Quality of Packets

The RTP Toolbox™ MOS call quality reporting is based on the Telchemy VQMon/SA application that reports call quality estimates including Listening and Conversational Quality MOS scores - MOS-LQ, MOS-CQ , and Listening and Conversational Quality R factors - R-LQ, R-CQ. Estimates are based on the ITU G.107 E Model which was based on a transmission planning tool developed and using opinion models in ETSI (technical report ETR 250). These models considered the entire Ear-Mouth path and all relevant conditions such as end-to-end level, echo, side tone, and frequency characteristics of the various path segments.

The E Model uses a computational method that includes factors such as noise, signal level, loudness ratings, impairments, delay, codec type, and even network type to derive a quality score. Over time and based on experience with subjective and objective measurements, the E Model's R-Factor score was mapped to an equivalent Mean Opinion Score (Excellent to Bad). Scoring includes consideration for the type of subjective test used for scoring. Passive/listening or active/conversational tests produce slightly different scores.

For IP networks, the score assumes ideal conditions outside the IP cloud and bases the scores on the relevant IP impairments such as packet loss, latency, jitter, and even when these impairments occur over the duration of the call.

Codec Name MOS-LQ MOS-CQ MOS-PQ R-LQ R-CQ VQMon-Nom MOS VQMon-Nom R factor
G.711 µ-Law 4.2 4.18 4.45 93 92 4.2 93
G.711 A-law 4.2 4.18 4.45 93 92 4.2 93
G.722 3.91 3.91   96 95 3.91 96
G.722.1 (32 K) 4.04 4.01   100 99 4.09 102
G.722.1 (24 K) 3.91 3.91   96 95 3.98 98
G.729A/G.729AB 3.91 3.88 3.8 82 81 3.91 82
GSM-FR 3.57 3.53 3.63 73 72 3.57 73
GSM HR 3.53 3.53 3.53 72 72 3.53 72
GSM EFR 4.16 4.16 4.16 91 91 4.16 91
G.726-40k 4.16 4.14 4.13 91 90 4.16 91
G.726-32k 4.04 4.01 3.89 86 85 4.04 86
G.726-24k 3.35 3.3 3.52 68 67 3.35 68
G.726-16k 2.82 2.77 3.2 57 56 2.82 57
G.726-40k with VAD 4.16 4.14 4.13 91 90 4.16 91
G.726-32k with VAD 4.04 4.01 3.89 86 85 4.04 86
G.726-24k with VAD 3.35 3.3 3.52 68 67 3.35 68
G.726-16k with VAD 2.82 2.77 3.2 57 56 2.82 57
AMR NB 7.95k
(requires additional license)
3.69 3.65 3.7 76 75 3.69 76
EVRC
(requires additional license)
3.94 3.94   83 83 3.94 83
EVRCB
(requires additional license)
3.98 3.98   84 84 3.98 84
SMV 3.61 3.57   74 73 3.88 81
Speex WB 4.14 4.16   106 105 4.16 106
Speex NB 4.14 4.16   91 90 4.16 91
iLBC 13.3k 3.88 3.84 3.79 81 80 3.88 81
iLBC 15.2k 3.95 3.91 3.82 83 82 3.95 83
  • G.711

    PCM has been the standard for digital voice transmission in telephony since 1972. 8 bit compressed pulse code modulation (PCM) samples at 8000 samples/second with 8 bits per sample. (64 kbit/sec) The two algorithms defined in the standard are µ-Law (North America & Japan) and A-law (used in Europe and the rest of the world). Both are logarithmic, but A-law was specifically designed to be simpler for a computer to process.

  • G.722

    G.722[1] is a ITU-T standard 7 kHz wideband speech codec operating at 48, 56 and 64 kbit/s. It was approved by ITU-T in November 1988. Technology of the codec is based on sub-band ADPCM (SB-ADPCM). G.722 sample audio data at a rate of 16 kHz (using 14 bits), double that of traditional telephony interfaces, which results in superior audio quality and clarity.

  • G.729

    Annex A and Annex B Voice encoding using CS-ACELP (Conjugate-Structure Algebraic Code Excited Linear Prediction) 8 kbps, is the lowest bit rate ITU-T standard with toll quality. Frame size is 10ms with a 5ms look ahead. Annex A is a low-complexity version of the G.729 standard. Annex B defines VAD/CNG/DTX (Voice Activity Detection/Comfort Noise Generator/Discontinuous Transmission) for G.729 and G.729A.

  • GSM-FR

    GSM-FR is a Full Rate speech coder standardized by the European Telecommunications Standards Institute (ETSI) for compressing toll quality speech (8000 samples / second) and was the first digital speech coding standard used in GSM digital mobile phone systems. The coder has a bit rate of 13 kbps.

    This coder uses the principle of Regular Pulse Excitation-Long Term Prediction-Linear Predictive coding. The coder works on a frame of 160 speech samples (20 msec), and no look ahead is required. So the algorithmic delay for the coder is 20 msec.

  • GSM HR
    GSM HR 6.20 operates with sampling frequency of 8000 samples/sec. This codec outputs the frames of size 14 Bytes at every 20msec which puts the bit rate of encoder at 5.6kbps. Codec supports Voice Activity Detection (VAD) to allow saving of bandwidth.

  • GSM EFR
    GSM-EFR (6.60) is an improved and hence the Extended version of GSM-FR(6.10) codec. With sampling frequency of 8000 samples/sec and frame size of 31 bytes/20 msec it achieves the bit rate of 12.2kbps. Codec supports Voice Activity Detection (VAD) to allow saving of bandwidth.

  • G.726 (ADPCM)

    This is an ADPCM (Adaptive Differential Pulse Code Modulation). Originally, a half-rate alternative to ITU-T G.711 and includes both the G.721 and G.723 standards. G.726 compresses by converting between linear, A-law (used in Europe) or µ-Law (used in the U.S and Japan) PCM and 40, 32, 24 or 16 kbps.

  • G.726 with VAD

    This is an ITU-T Adaptive differential pulse code modulation (ADPCM) voice codec, which transmits at bit rates of 16, 24, 32, and 40 kbps. It supports Voice Activity detection and generates SID packets during Silence Period. ADPCM provides the following functionality:

    • Voice mail recording and playback, which is a requirement for Internet voice mail.
    • Voice transport for cellular, wireless, and cable markets.
    • High voice quality voice transport at 32 kbps.

  • G.726 with Voice Activity Detection (ADPCM)

    This is an ITU-T Adaptive differential pulse code modulation (ADPCM) voice codec, which transmits at bit rates of 16, 24, 32, and 40 kbps. It supports Voice Activity detection and generates SID packets during Silence Period. ADPCM provides the following functionality:

    • Voice mail recording and playback, which is a requirement for Internet voice mail.
    • Voice transport for cellular, wireless, and cable markets.
    • High voice quality voice transport at 32 kbps.

  • AMR NB (requires additional license)

    AMR is the 3GPP mandatory standard codec for narrowband speech and multimedia messaging services over GSM and evolved GSM (WCDMA, GPRS and EDGE) networks. Designed to provide transcoder free connectivity between GSM, US-TDMA and Personal Digital Cellular networks (Japan).

    AMR operates at eight bit rates in the range of 4.75 to 12.2 kbps and was specifically designed to improve link robustness.

  • EVRC (requires additional license)

    For EVRC codec type three rates are provided (1/8, ½ and 1). Default 1/8 and 1 are selected as the minimum rate & maximum rate. Minimum rate should be less than or equal to maximum rate. There is option to select RTP packet format between Header Free Format and Bundled Format. By default Bundled Format is set.

  • EVRCB (requires additional license)

    For EVRCB codec type four rates are provided (1/8, ¼,½ and 1). Minimum rate should be less than or equal to maximum rate. By default 1/8 and 1 are selected as the minimum rate & maximum rate. There is option to select RTP packet format between Header Free Format and Bundled Format. By default Bundled Format is set.

  • SMV

    The Selectable Mode Vocoder (SMV) [2] compresses each 20 milliseconds of 8000 Hz, 16-bit sampled speech input into output frames of one of the four different sizes: Rate 1 (171 bits), Rate 1/2 (80 bits), Rate 1/4 (40 bits), or Rate 1/8 (16 bits). SMV is the preferred speech codec standard for CDMA2000, and will be deployed in third generation handsets.

  • Speex WB

    This Codec has a sampling rate of 16000 samples/sec, which makes it a wide band codec. This codec supports different codec options such as Sampling Rate, Variable Bit Rate, Voice Activity Detection and Perceptional Enhancement.

  • Speex NB

    Based on CELP Narrowband (8 kHz), Open source codec targeted for VoIP and file-based applications

  • iLBC Codec

    iLBC (internet Low Bitrate Codec) is a narrow band speech codec that operates at either 13.33 kbit/s with an encoding frame length of 30 ms and 15.20 kbps with an encoding length of 20 ms. Companies that are using iLBC in their commercial products include:

    • Applications/Soft phones: Skype, Nortel, Webex, Hotsip, Marratech, Gatelinx, K-Phone, XTen;
    • IP Phones: WorldGate, Grandstream, Pingtel;
    • Chip: Audiocodes, TI Telogy, LeadTek, Mindspeed.

    The 13.33 kbps rate 30ms frame encodes packets of 399 bits, (50 bytes) and is designated in RTP Toolbox as iLBC_13_33.

    The 15.2 kbps 20 ms frame creates packets of 303 bits, (38 bytes). This is labeled iLBC in RTP Toolbox. The basic quality is higher than G.729A.

Some Definitions:

R-Factor Quality score based on various end point and network parameters. Includes codecs, packet loss, and delay.
Conversational R-Factor The voice quality metric that measures voice quality based on transmission delay, burst packet loss, and burst loss recency.
Listening R-Factor The voice quality metric based only on burst packet loss and codec selection.
MOS-LQ Mean Opinion Score based on listening quality. Does not consider recency or delay. ITU-T P.862 Listening Quality implementations.
MOS-CQ Mean Opinion Score based on conversational quality. Includes recency and delay effects.
MOS-PQ ITU-T P.862 normalized raw quality score.
MOS-Nom Nominal quality or maximum score for the codec selected. Similar to the G.107 E-model defaults
Recency A time factor used to weight scores based on the time from a burst packet loss to the end of the call or next packet loss event.


Applications

Media Gateway Testing using RTP ToolBox™

  • Complete G.168 Compliance Testing (All 13 Tests)

    G. 168 Test Name Supported?
    Test 1: Steady state residual and returned echo level test Yes
    Test 2A:Convergence test with NLP enabled Yes
    Test 2B: Convergence test with NLP disabled Yes
    Test 2C: Convergence test in the presence of background noise Yes
    Test 3: Performance under conditions of double talk Yes
    Test 4: Leak rate test Yes
    Test 5: Infinite return loss convergence test Yes
    Test 6: Non-divergence on narrow-band signals Yes
    Test 7: Stability test Yes
    Test 8: Non-convergence of EC on SS5/SS6/SS7 tones Yes
    Test 9: Comfort noise test Yes
    Test 10A: Canceller operation on the calling station side Yes
    Test 10B: Canceller operation on the called station side Yes
    Test 11: Tandem echo canceller test For further study
    Test 12: Residual acoustic echo test For further study
    Test 13: Performance with low bit rate coders Under study
    Test 14: Performance with V-series low-speed modems Optional
    Test 15: PCM offset test Yes

  • Voice Quality Testing using PESQ, PAMS and PSQM

  • Codec Testing and Verification

G.168 Compliance Test for EC Within ATA

G.168 Tests which can be preformed on an ATA using RTP ToolBox™

G. 168 Test Name Supported?
Test 1: Steady state residual and returned echo level test Yes
Test 2A:Convergence test with NLP enabled Yes
Test 2B: Convergence test with NLP disabled Yes
Test 2C: Convergence test in the presence of background noise No
Test 3: Performance under conditions of double talk No
Test 4: Leak rate test Yes
Test 5: Infinite return loss convergence test No
Test 6: Non-divergence on narrow-band signals Yes
Test 7: Stability test Yes
Test 8: Non-convergence of EC on SS5/SS6/SS7 tones No
Test 9: Comfort noise test No
Test 10A: Canceller operation on the calling station side No
Test 10B: Canceller operation on the called station side Yes

Voice Quality Testing in VoIP

  • Using GL's RTP Toolbox™, TCL Scripts, and VQT Algorithm

    The RTP ToolBox™ application provides multiple features to perform Voice Quality Testing over IP networks. The software can send and record the voice files over the "network under test" either manually through GUI or programmatically through client-server scripts. Different codec types, impairments, jitter buffer, and latency events can be set to simulate the necessary network conditions. Complete automation can be accomplished using the TCL client scripting. The following applications are described in detail.

    • Manual testing
    • Testing VoIP handset directly
    • Automated testing

    Click here to know more details.

  • Using GL’s VQuad™ with other VoIP products

    The VQuad™ software is used for sending and recording the voice files across various networks. The Universal Telephony Adapter (UTA) supports interfacing between the telephone and the handset, allowing audio to be injected and recorded through the phone and across the given network. The VQT software is then used to perform PESQ, PAMS, and PSQM (+) measurements simultaneously, using the two voice files (reference and degraded files from VQuad™) and provides the algorithm results along with analytical results, in both a graphical and tabular format.

    Click here to know more details.


Buyer's Guide:

Please Note1: The XX in the Item No. refers to the hardware platform, listed at the bottom of the Buyer's Guide, which the software will be running on. Therefore, XX can either be ETA or EEA (Octal/Quad Boards), PTA or PEA (tProbe Units), UTA or UEA (USB Units), HUT or HUE (Universal Cards), and HDT or HDE (HD cards) depending upon the hardware.

Item No. Item Description
PKB100 RTP ToolBox™ Application
PCD103 AMR Codec for RTP ToolBox™ (requires additional license)
PCD104 EVRC Codec for RTP ToolBox™ (requires additional license)
PCD105 EVRC-B Codec for RTP ToolBox™ (requires additional license)
PCD106 EVRC-C Codec for RTP ToolBox™ (requires additional license)
PKB105 G.168 Echo Canceller Test Compliance Suite
PKB110 RTP ToolBox™ Client-Server Application (C++, TCL)
  Related Software
PKS100 PacketGen™ with PacketScan™
PKS101 SIP Core (additional)
PKS102 RTP Soft Core for RTP Traffic Generation (additional)
PKS110 Packet H. 323
PKS120

PKS121

PKS122

PKS123

PKS124

PKS125

PKS130
MAPS™ SIP Emulator

MAPS™ SIP Conformance Test Suite (Test Scripts)

MAPS™ MEGACO Emulator

MAPS™ MEGACO Conformance Test Suite (Test Scripts)

MAPS™ MGCP Emulator

MAPS™ MGCP Conformance Test Suite (Test Scripts)

MAPS™ SIGTRAN Emulator
PKV100 PacketScan™ (Online and Offline)
PKV101 PacketScan™ - Offline
PKV105 SIGTRAN
IPN010 IPNetSim™ - 100Mbps of through bandwidth
IPN100 IPNetSim™ - 1Gbps of through bandwidth
IPN400 IPNetSim™ - 1Gbps w/ 4 links through bandwidth
VQT004 VQT with PESQ, PAMS and PSQM
VQT002 VQT with PESQ only
XX062 Echo Path Delay/Loss Simulation Software
EMU037 Echo Measurement Utility (EMU) Software
  Related Hardware
HDT001/HDE001 Legacy HD T1 or E1 (PCI) Cards with Basic Analyzer Software
UTE001 USB Based T1 or E1 Analyzer Unit with Basic Analyzer Software

Specifications are subject to change without notice.

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