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Home > VoIP Analysis and Simulation
RTP ToolBox™ Software Ver 2.08 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 | 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
RTP ToolBox™ Testing Applications
Main Features
- Supported Codec's:
- G.711 (µ-law/A-law - 64kbps)
- G.729, G.729B (8 kbps)
- G.726 (40/32/24/16 kbps)
- GSM (13.2 kbps)
- AMR (4.75kbps, 5.15kbps, 5.9kbps, 6.7kbps, 7.4kbps, 7.95kbps, 10.2kbps, 12.2 kbps)
- EVRC (Rates - 1/8, ˝ and 1)
- 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))
- G722
- G722.1 (24 kbps and 32 kbps Wideband)
- G.711 with VAD
- ISAC - (An optional Codec, must be purchased as a separate dongle extension)
- AMR ( Wideband codec - 6.60kbps, 8.85kbps, 12.65kbps, 14.25kbps, 15.85kbps, 18.25kbps, 19.85kbps, 23.05kbps, 23.85 kbps)
For more comprehensive information on the above codecs click here.
- Set the RTP traffic properties (payloadtype, 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 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
- Quality Metrics with R-Factor and MOS Factors, Jitter Buffer Statistics, Degradation Factor, Burst Metrics, and Delay
Metrics are graphically represented
- Supports Client-Server functionality (requires additional license)-C++, & TCL clients
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.
Click to Enlarge
Click to Enlarge
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 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
Click to Enlarge
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).
Click to Enlarge
Jitter Buffer Statistics, Quality Metrics (R Factor & MOS), Degradation Factor, Burst Metrics, Delay 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 Delay Metrics, and Burst Metrics.
Click to Enlarge
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.
Click to Enlarge
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.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 |
| 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 |
| AMR NB 7.95k |
3.69 |
3.65 |
3.7 |
76 |
75 |
3.69 |
76 |
| Speex NB 5.95k |
2.92 |
2.87 |
3.26 |
59 |
58 |
2.92 |
59 |
| 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.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.
- G.726
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.
- AMR NB
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.
- 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 Note: 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 DPT, DPE, PCT, PCE, HDT, HDE, DLT, DLE or UTE
depending upon the hardware.
Specifications are subject to change without notice.
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