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RTP Toolbox™
RTP Packet Testing & Simulation Tools

RTP ToolBox™ Software Ver 6.6.7  | Download Now!

Overview | Main Features | Voice Codec Options | 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


GL's RTP ToolBox™ (PKS102) 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.

pdf Product Brochure
pdf Quick Install Guide
pdf Quick Verification Guide
pdf User's Guide
Supported Codecs
pdf Whitepaper
pdf View Presentation

RTP ToolBox™ Testing Applications

Main Features


  • Create RTP sessions & Auto scan incoming RTP sessions; supports IPv6 addressing.
  • Can run on any PC with Windows® 7 /8 (32 bit and 64 bit) OS
  • G.168 testing for echo cancellation equipment.
  • User-defined impairments: latency, packet loss, out of sequence, and duplicate packets.
  • Talk and play to speaker options using PC sound card.

Call Generation

  • 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 & Reception.
  • Multiple frame interval or Packetisation Time supported for almost all codec s.
  • Generation / Detection of in-band and out-of-band Digits / Tones (DTMF, MF, user-defined, etc) / Events per RFC-2833 & RFC-4733.

Traffic Handling

  • Set the RTP traffic properties (payload type, codec) and impairments during auto-scan.
  • Sending and recording of voice files (.glw) with a synchronous Tx/Rx option.
  • Set delay and attenuate for incoming RTP traffic.


  • Monitoring RTP streams and captured data using scalable Oscilloscope and Spectrum Analyzer.
  • Detailed statistical information of RTP and RTCP packets.
  • Quality Metrics with MOS (G.107 based E-model/R-Factor), jitter buffer statistics, degradation factor, and burst metrics are graphically represented.

Voice Codec Options

The Call Generation (Dial) & Call Reception features provides various codec parameters in the TX/RX profiles during negotiation.

  • Allows to specify a desired voice payload type to each codec for sending and receiving payload;
  • Sampling rate of the codec is displayed for the selected codec.
  • Comfort noise generation is supported for A-law, µ-law and G.726 codecs for sending and receiving payload.
  • Allows to set the buffer used for delayed packets that arrive at receiving end (both static and dynamic jitter buffers are supported)
  • Allows to set QoS (Type of Service) properties such as precedence, delay, throughput and reliability values to the outgoing stream

RTP ToolBox™ supports the following codecs:

  • G.711 (A-law / Mu-law - 64kbps), G.711 App II (A-Law and Mu-Law with VAD Support)
  • G.722 (64 kbps) , G.722.1 (32 kbps and 24 kbps)
  • G.729, G.729B (8 kbps)
  • G.726 , G.726 (40/32/24/16 kbps with VAD)
  • GSM 6.10 FR (13.2 kbps), GSM-HR (rate – 5.6kbps)
  • GSM-EFR ( 12.2kbps, packet time fixed at 20ms)
  • SPEEX, SPEEX_WB  (packet time fixed to 20msec)
  • iLBC, iLBC_13_33
  • SMV (Modes - 0, 1, 2 and 3- Available if licenses are provided or owned, please call GL)
  • AMR (4.75kbps, 5.15kbps, 5.9kbps, 6.7kbps, 7.4kbps, 7.95kbps, 10.2kbps, 12.2 kbps), AMR WB  (optional codec)
  • EVRC ( 1/8, ½ and 1) , EVRC0 (optional codec)
  • EVRC_B ( 1/8, ¼, ½, and 1), EVRCB0 (optional codec)
  • EVRC_C (optional codec)

Visit Voice Codecs webpage for more comprehensive information.

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 and POLQA 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


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, 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.

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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Speech Quality Metrics (R Factor & MOS)

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. An typical estimate of the MOS and R-Factor scores for each codec is available in

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


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 and POLQA

  • 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 and POLQA 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

Optional Codec – AMR – Narrowband (requires additional license)


Optional Codec - EVRC (requires additional license)


Optional Codec – EVRC-B (requires additional license)


Optional Codec – EVRC-C (requires additional license)


Optional Codec – AMR - Wideband (requires additional license)


Optional Codec  - EVS (requires additional license)

PCD109 Optional Codec  - Opus (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)
PKS103 RTP IuUP Softcore



PKS106 RTP Video Traffic Generation

RTP Voice Quality Measurements






MAPS™ SIP Emulator

MAPS™ SIP Conformance Test Suite (Test Scripts)


MAPS™ MEGACO Conformance Test Suite (Test Scripts)

MAPS™ MGCP Protocol Emulation with Conformance Test Suite

PKV100 PacketScan™ (Online and Offline)
PKV101 PacketScan™ - Offline
IPN502 IPNetSim™ - 1G – MultiStream – Rack System
IPN504 IPNetSim™ - 10G – MultiStream – Rack System
IPN505 IPNetSim™ - 1G Tablet
VQT002 VQT with PESQ only
VQT006 VQT w/ POLQA Server License
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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