PacketScan™ Software Ver 2.0.58 is Now Available! Download it here
Overview |
Main Features |
Supported Protocols|
Real-Time VoIP Traffic Analysis
Packet Data Analysis |
Speech Quality of Packets |
Other VoIP Products
Frequently Asked Questions |
Application Notes |
Buyer's Guide
Click the image to view Live Demo of PacketScan™
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Overview
PacketScan™ is a real-time VoIP analyzer that captures live IP traffic, segregates them into SIP/H323 calls and
collects statistics about the calls. Hundreds of calls can be monitored in real-time including detailed analysis of selected voice band
streams. Applications include testing of IP phones, Gateways, IP Routers and Switches, and Proxies.
Users can listen/record VoIP calls in real-time; perform power, frequency, spectral, tone and digit analysis with ease and precision; get an exact picture of
QOS (quality of the service) and the technical adherence (adherence to the protocols specified by the standardizing authority) of the system under test.
Detailed call statistics such as packet loss, gap, jitter, delay, RTP performance statistics, R-factor & MOS scores, and unparalleled voice band statistics can
be monitored simultaneously. Sophisticated filters permit zooming and recording of specific calls of interest.
It allows to record all or filtered traffic into a trace file.
PacketScan™ includes Virtual Packet Analysis (VPA) and Packet Data Analysis (PDA)/ Traffic Analyzer (TA) views.
PacketScan™ further supports the following three views in Traffic Analysis or Packet Data Analysis (PDA) window:
- Summary View (Call Quality Matrix) displays complete summary of SIP/ H323 /MEGACO / RTP/ call information in graphical format along with a summary of alerts.
- Detail View (RTP Diagnostic View) displays packet by packet statistics for particular call information in tabular format
These interfaces allow users to define parameters such as E-Model based MOS and R-Factor score, VQmon settings, & Dynamic
Payload Mapping. It also supports real-time digit capturing like DTMF, MF and user-defined digits/tones.
- Registration Summary View displays statistics and status of the SIP registration process, an active registration graph, and registration trace
graph of each SIP registration.
What sets apart PacketScan™ is its ability to collect vital statistics about calls for theoretically infinite time. The ability of PacketScan™ to capture
data is limited only by the hard disk capacity of the PC.
Main Features
- Monitor progress of up to 500+ simultaneous calls with bidirectional RTP traffic
- Supported protocols - SIP (SIP Session Initiation Protocol 3261), Megaco3525, Megaco3015, MGCP, H323/H225, and RTP.
- Supports decoding of MAC, IP, SIP, UDP, TCP, RTP, RTCP, Megaco, T.38 (Fax over IP), and SMPP (Short Message Peer to Peer Protocol).
- PacketScan™ supports standard codecs such as
- G.711 (µ-law and A-Law - 64 Kbps)
- G.711 App II (A-law and µ-law with VAD)
- G.726 (40/32/24/16 kbps)
- GSM (13.2kbps)
- G729 (8kbps)
- G729B (8kbps)
- G.722
- G.722.1 (24 kbps and 32 kbps Wideband)
- AMR (Wideband codec - 6.60kbps, 8.85kbps, 12.65kbps, 14.25kbps, 15.85kbps, 18.25kbps, 19.85kbps, 23.05kbps, 23.85 kbps)
- AMR (Narrow band codec - 4.75kbps, 5.15kbps, 5.9kbps, 6.7kbps, 7.4kbps, 7.95kbps, 10.2kbps, 12.2 kbps)
- ILBC (15.2kbps and 13.33kbps)
- SPEEX (Wideband)
- SPEEX (Narrowband)
- ISAC (An optional Codec, must be purchased as a separate dongle extension)
- EVRC (Rates - 1/8, ½ and 1)
- EVRCB (Rates – 1/8, ½ and 1)
- SMV (Modes - 0, 1, 2 and 3 - Available if licenses are provided or owned, please call GL)
- H.263+ providing video capture and videoconference monitoring capability.
For more comprehensive information on these codecs click here.
- Full RTP Analysis with audio capture/playback supported for all common codecs
- 1 GB circular buffer with capture and view filters
- Decode AMR in all packing formats, and G.726 RTP & AAL packing types
- VPA View provides Summary, Detail, Hex dump, and Call Detail Records View of the captured traffic.
- Summary View displays Frame number, Time, Frame Length, Error, Source TCP/IP/UDP Address, Destination TCP/IP/UDP address, Packet Type,
and other SIP/RTP call details in a tabular format.
- Detail View displays decodes of user-selected frame from MAC, IP, UDP/TCP, SIP, RTP, RTCP, H323 etc in detail. Further details of an individual (RTP)
session of a call can also be obtained using Traffic Analyzer (TA) or the Packet Data Analysis (PDA) interface.
- Statistics View displays statistics based on frame count, byte count, frames/sec, bytes/sec etc for the entire capture data. Statistics can be
obtained for any fields or parameters in the protocol.
- Hex Dump View displays raw frame data as hexadecimal.
- Exports Summary View information to a comma delimited file for subsequent import into a database or spreadsheet.
- Capability to export detail decodes information to an ASCII file.
- Search and filtering capabilities for both real time as well as offline analysis.
- Ability to configure sipprot.ini file for customization of decoding options.
- Can be deployed as a centralized monitoring system with associated relational or Oracle database.
PDA Main Features
- Packet Data Analysis (PDA) interface displays call information in graphical format (Traffic Analyzer Call Quality Matrix- Summary View) as well as in
tabular format (RTP Diagnostic - Detail View).
- Call capturing based on call agents or trigger actions such as MOS, R-factor, jitter, packet loss, duplicate packets, or called / calling numbers
(SIP/H323/Megaco).
- Provision for H.263+ video capture and video conference monitoring capability
- R factors/MOS is supported for audio codecs such as Mulaw, Alaw, G726 (40, 32, 24, 16 kbps), GSM610, G729, G729B, AMR, ILBC (20, 30 msec),
SPEEX, and G711 application II.
- User can get real-time call trace information based on H225 calls
- Calculates minimum, maximum, and average Round Trip Delay (RTD) values for SIP calls
- Call Quality Of Service (QOS) for all calls with E-Model based (G.107) Mean Opinion Score (MOS) and R-factor with
individual and summary statistics presented in graphical and tabular formats
- Graphs are provided for key values to give a pictorial representation of the statistics, there by allowing the user to monitor
the network with ease; some of the graphs available are – Active Calls, Average Jitter, E-Model MOS/R-Factor/Packets Discarded, RTP
Packets Summary, ladder diagram for T.38 based fax calls and call signaling
- Displays summary of signaling and audio parameters of each call in call summary.
- A host of counters gives the user an instantaneous snapshot of the VoIP traffic on the network
- Calls and sessions are classified as active, completed or failed giving the user an idea about the calls and its status in the network
- Supports saving the selected calls from traffic analyzer into *.hdl or *.pcap formats.
- Provides the registration summary of each SIP registration including the user agent, registrar, status, registered time, expiry time, time to live,
and remaining time.
- Provides graphical view of the active registrations and registration trace of each registration.
- Generates alert summary when particular vital parameters go beyond a specified value.
- Real-time audio/video monitoring of RTP streams using Audio Playback, Record Video, and Write to File features.
- Its ability to monitor / record audio and video data of a session to files (in QuickTime *.qt format), allows users to perform powerful video analysis.
The captured VoIP calls with video can be played back using 3rd party VLC Viewer application.
- Audio Write to file allows user to save the RTP streams as wave files, for later playback or for analysis using more sophisticated audio manipulating tools.
Recommended PC
- Windows 2000/XP/Vista Operating System
- 2 GHZ, 512 MB RAM, 40 GB Hard Drive, 10/100/1000 Ethernet Port, Parallel or USB port for License, Sound Card, Headphones and Microphone
Supported Protocols
A brief overview of the protocols supported by PacketScan™ is given below:
- Session Initiation Protocol (SIP) - RFC 3261 and 2543
SIP can be used with other IETF protocols to build complete multimedia architecture. Typically, these architectures will
include protocols such as the Real-time Transport Protocol (RTP) (RFC 1889) for transporting real-time data and providing
QoS feedback, the Real-Time streaming protocol (RTSP) (RFC 2326) for controlling delivery of streaming media, the Media
Gateway Control Protocol (MEGACO) (RFC 3015) for controlling gateways to the Public Switched Telephone Network (PSTN),
and the Session Description Protocol (SDP) (RFC 2327) for describing multimedia sessions.
- Media Gateway Control Protocol (MGCP) - RFC 2705/3435 (3991)
MGCP is a protocol for controlling Voice over IP Gateways (or Call Agent endpoints) from external call control elements. It assumes a
call control architecture where the call control "intelligence" is outside the gateways and handled by external call control
elements. The Call Agent can create, modify and delete connections in order to establish and control media sessions with
other multimedia endpoints. Also, the Call Agent can instruct the end points to detect certain events and generate
signals. The endpoints automatically communicate changes in service state to the Call Agent.
- Media Gateway Control (MEGACO) - RFC 3525 and 3015
Megaco, also known as H.248 is a signaling protocol, is used between Media Gateway and Media Gateway Controller
(Call Agent). Megaco/H.248 is architecturally quite similar to MGCP, however Megaco/H.248 supports a broader range
of networks.
- H.323
H.323 provides the foundation for audio, video and data communication on packet based IP network. H.323
specifies functions provided by other ITU_T as well as IETF standards under one umbrella. First, ITU_T- H225 provides call setup/disconnect
and terminal to gatekeeper signaling. ITU_T-H245 protocol adds terminal control functions that are used to negotiate terminal capabilities,
channel usage and other end-to-end functions. The IETF RTP and RTCP provide information transport as well as session management.
Finally, ITU_T standard voice and video encoding algorithm provide the analog to digital conversion and signal compression required for
bandwidth optimization.
PacketScan™ (Real-Time VoIP Traffic Analysis/Monitoring Tool)
Summary, Details and HEX/ASCII Views
The PacketScan™ main screen allows real-time as well as offline protocol analysis. The analyzer displays Summary, Detail, and Hex dump
view in different panes. The Summary View displays various information such as Frame Number, Time, Length, Message Types, IP source and destination
address and so on. User can select a frame in Summary View to analyze and decode each frame in the Detail View. The Hex dump view displays the
frame information in HEX and ASCII octet dump format.
Screen Shot of PacketScan™ Main Window
Real-time and Offline Analysis
Users can capture and analyze packets through real-time analysis or analyze the recorded data in offline mode. All
captured or filtered traffic can be recorded into a trace file. The recorded trace file can be analyzed offline and exported to
ASCII file, or printed.
Screen Shot of Real-time Analysis
Call Detail Records & Statistics View
Call Detail Records View displays important call specific parameters like call status, type, call identifier, duration of call, CRV, release
cause, parties involved and more.
Statistics are an important feature available in PacketScan™ and can be obtained for all frames both in real-time
as well as offline mode. Various statistics can be obtained to study the performance and trend in the VoIP network, based
on protocol fields and different parameters such as User Type (Key/Total/Field) , Statistic type (Frame count, Byte count,
Frames/Sec) and patterns like Range List, Wild card.
Screen Shot of Define Statistics
Filtering and Search
Users can capture and analyze packets using real-time analyzer and record all or filtered traffic into a trace file. The
recorded trace file can then be analyzed offline and exported to ASCII file, or printed. Filtering and search capability adds
a powerful dimension to the SIP analyzer. This feature can isolate required frames from original frames in real-time/offline.
It allows real-time filtering based on parameters set in Data Link layer, MAC layer, IP, TCP/UDP and more. The offline filter
allows filtering based on Frame Number, Time, Length, Message Types, and so on. Similarly, search capability helps user
to search for a particular frame based on specific search criteria.
Screen Shot of Real-time Capture filter
Screen Shot of Offline View Filter
Enhanced Trace Saving Options
Users can control the captured trace files by saving the trace using different conventions such as trace files with
user-defined prefixes, trace file with date-time prefixes, and slider control to indicate the total number of files, file size, frame
count, or time limit. This feature also allows the captured frames to be saved into a trace file based on the filtering criteria set
using display filter feature.
Configuring INI Files
Users can edit the SipProt.ini INI files to customize the decoding options. The following parameters can be edited as per
user requirements
- [#RTP_PAYLOAD_RFC4733]: This header indicates the RFC to be followed for out of band events
- [#RFC2833]: This indicates out of band payload type
- [#RFC2190]: This field indicates payload type for H.263
- [#WITHOUT_PAYLOAD]: This field indicates the static payload range
Save/Load All Configuration Settings
Protocol configuration window provides a consolidated interface for all GUI and protocol settings required in the analyzer. This includes various
options such as protocol selection, startup options, filter/search criteria and so on.
All the configuration settings done in any of these options can be saved to a file, loaded from a configuration file, or user may
just revert to the default values using the default option.
Packet Data Analysis (PDA) / Traffic Analysis (TA)
PacketScan™ provides in-depth real-time and post-process data investigation. The PDA view assists in
any comparisons that are to be made between the two RTP sessions. Traffic Analysis Detail View allows the user to have a
detail look at the two (or one) RTP sessions that are part of a single call. This distinction assists in any comparisons that
are to be made between the two sessions. Here each frame of the selected session is dissected and its contents are
displayed in a tabular form for easier viewing and comparisons. Vital aspects from the RTP frame needed for close analysis
are included provides in tabular and graphical formats.
Screen Shot of Summary & Detail Window
Information provided within the Traffic summary and detailed views include:
Real-Time Digit Capturing
PacketScan™ provides a means for digit examination and capturing. An easy-to-read window indicates the direction of the
captured digits along with the duration, power and frequencies. DTMF, MF and User-defined digits/tones are fully supported.
Screen Shot of Real-Time Digit Capturing Window
RTP (Audio) Listen and Record Applications (Play To Speaker & Write To File)
Play to Speaker
Play To Speaker application allows the user to play the RTP streams of a call to the PC speaker using a soundcard. A host of options
such as Jitter Buffer settings, As Is, Audio Mixing and so on are available for users to play a live call in real-time or play captured voice
files.
Screen Shot of Play Jitter Options
Write To File
Similar to Play to Speaker application, various options are provided for the user to save the captured file in a required format, and use
the files with voice quality analysis software to investigate more about the quality of voice in the network. This application write or records
the RTP stream to a file in *.wav format. (Examine both directions using Waveform Viewer)
Screen Shot of Write to File
Record Video
This feature will allow the user to record audio and video data of a session to a file in QuickTime format. PacketScan™ can monitor video
calls and display both audio and video RTP streams in summary view. Video calls will be marked with symbol "V" at the left corner as shown
in the figure below.
The video call can be recorded to a file in QuickTime format (*.qt) by selecting a SIP/Auto Detected RTP call. PacketScan™ may be installed
in either of the system to record the video data as depicted in the image below.
The recorded video data in Quick Time Format (*.qt) can be viewed by VLC Viewer (3rd party application).
Record video option is available for both Auto Detected RTP Calls and SIP Calls. Supported Video Codecs are:
- H263+
- H263++ CIF 190 kbps
- H263++ CIF 350 kbps
- H263++ CIF 512 kbps
- H263++ QCIF 128 kbps
- H263++ QCIF 64 kbps
- H263++ QCIF 80 kbps.
H.263 is a video codec designed by the ITU-T as a low-bitrate encoding solution for videoconferencing.
QCIF - Quarter Common Intermediate Format, a videoconferencing format that specifies data rates of 30 frames per second (fps), with
each frame containing 144 lines and 176 pixels per line.
CIF - A video format used in videoconferencing systems that specifies a data rate of 30 frames per second (fps), with each frame
containing 288 lines and 352 pixels per line.
Registration Summary
Packet Data Analysis is enhanced to display the SIP registration information in tabular as well as graphical format in Registration Summary. This interface
displays the SIP registration information in a tabular format which includes user agent, registrar, registered time, status, and so on for each user agent. In
addition, it displays the active registration graph of the entire registration summary and provides the trace display of each registration.
Screenshot of Registration Summary
Include graphs that display the active registration and registration trace message sequences of registered calls at the bottom of the registration
summary.
Other VoIP Products
VoIP Network Testing w/ GL's Media Gateway
The DCOSS MG (Media Gateway) system supports dual redundant Ethernet trunks for traffic and signaling testing of VoIP and IP networks.
The DCOSS / MG combination provides user friendly bulk call generation and reception for various voiceband traffic types including voice, fax, and
modem. It can be connected to any IP Phone, softphone, VoIP PBX, or VoIP Network / Cloud. For more information click
here
PacketGen™ - VoIP Testing and Simulation Tool
GL's PacketGen™ application allows users to generate/accept thousands of VoIP calls with full media
stream manipulation. It is designed to function as a bulk call generator for stress testing and precise analysis of the VoIP network
equipment.
Message Automation & Protocol Simulation (MAPS) - SIP & Megaco Tester
Message Automation & Protocol Simulation (MAPS) application is designed for SIP and Megaco protocol conformance
testing.
MAPS – Megaco Tester - Provides MEGACO test library capable of testing
Media Gateways (MG) and Media Gateway Controllers (MGC). The test library includes test functions to generate MEGACO
messages, edit messages, simulates a variety of call flows, and control scenario.
The MAPS - MEGACO Conformance Suite tool is designed with 250+ test cases, as per specification of ETSI TS 102
374-2 (2004-11) standard. The test suite includes in-built scripts to process MEGACO valid and in-valid behaviors.
MAPS – SIP Tester - supports testing SIP proxy servers, Redirect servers,
Registrars, Registrants, and user agents such as SIP phones. Test cases include general messaging and call flow
scenarios for multimedia call session setup and control over IP network. Logging and pass/fail results are also reported.
The MAPS - SIP Conformance Suite tool is designed with 300+ test cases, as per SIP specification of ETSI TS
102-027-2 v4.1.1 (2006-07)) standard. Test cases verify conformance of actions such as registration, call control,
capability queries and messaging for registrars, proxies and redirect servers.
SIGTRAN Analyzer
GL's SIGTRAN protocol decode permits real-time analysis, call trace, capture, and filter of
SS7and ISDN signaling messages over IP protocol. This software is a companion to GL's
award winning SS7 and ISDN protocol analyzers.
RTP ToolBox™ - RTP Testing and Simulation Tool
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.
IPNetSim™
IPNetSim™ can simulate an entire IP network in a single box. All the conditions encountered in a real-time IP network
are simulated such as network latency, network delay variation (jitter), bandwidth, congestion, packet errors, bit errors
and other link impairments independently in both directions at speeds of up to 100 Mbps (or1Gbps per link). It can be
used to test IP end point devices such as (Gateways, IADs, IP phones, Soft phones, and so on) for voice, fax, data, or
multimedia transmission over IP.
Frequently Asked Questions
Application Notes
Buyer's Guide:
Specifications are subject to change without notice.
Back to VoIP Analysis and Simulation Index Page
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![Screenshot of Active Calls Graph](javascript:myOpen("images/packetscan_img26.gif")){kind=link}
![Screenshot of Average Jitter Distribution Graph](javascript:myOpen("images/packetscan_img1.gif")){kind=link}
![Screenshot of Call Graph](javascript:myOpen("images/packetscan_img28.gif")){kind=link}
![Screenshot of MEGACO Call Graph](javascript:myOpen("images/packetscan_img37.gif")){kind=link}
![Screenshot of RTCP Details Screen](javascript:myOpen("images/packetscan_img4.gif")){kind=link}
![Screenshot of Gap Graph](javascript:myOpen("images/packetscan_img5.gif")){kind=link}
![Screenshot of Jitter Graph](javascript:myOpen("images/packetscan_img6.gif")){kind=link}
![Screenshot of Gap Distribution Graph](javascript:myOpen("images/packetscan_img7.gif")){kind=link}
![Screenshot of Jitter Distribution Graph](javascript:myOpen("images/packetscan_img8.gif")){kind=link}
![Screenshot of Full Duplex Wave Graph](javascript:myOpen("images/packetscan_img9.gif")){kind=link}
![Screenshot of Call Summary – Signaling and Audio Parameters](javascript:myOpen("images/packetscan_img33.gif")){kind=link}
![Screenshot of Inband Events Screen](javascript:myOpen("images/packetscan_img10.gif")){kind=link}
![Screenshot of RTP Events Screen](javascript:myOpen("images/packetscan_img13.gif")){kind=link}
![Screenshot of E-Model Graph - R Factor, MOS, Packet discarded graphs in PDA Summary View](javascript:myOpen("images/packetscan_img2.gif")){kind=link}
![Screenshot of RTP Packets Graph](javascript:myOpen("images/packetscan_img3.gif")){kind=link}
![Screenshot of MOS Graph](javascript:myOpen("images/packetscan_img11.gif")){kind=link}
![Screenshot of Quality Factors](javascript:myOpen("images/packetscan_img12.gif")){kind=link}
![Screenshot of Quality Metrics](javascript:myOpen("images/packetscan_img29.gif")){kind=link}
![Screenshot of Burst Metrics](javascript:myOpen("images/packetscan_img30.gif")){kind=link}
![Screenshot of Delay Metrics](javascript:myOpen("images/packetscan_img31.gif")){kind=link}
![Screenshot if Jitter Buffer Statistics](javascript:myOpen("images/packetscan_img32.gif")){kind=link}
![Screenshot of VQmon Settings](javascript:myOpen("images/packetscan_img20.gif")){kind=link}
![Screenshot of Dynamic Payload Mapping](javascript:myOpen("images/packetscan_img21.gif")){kind=link}
![Screenshot of Triggers and Action Settings](javascript:myOpen("images/packetscan_img22.gif")){kind=link}
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![Screenshot of T.38 Analysis over IP](javascript:myOpen("images/packetscan_img27.gif")){kind=link}
![Screenshot of Registration Active Graph](javascript:myOpen("images/packetscan_img35.gif")){kind=link}
![Screenshot of Registration Trace](javascript:myOpen("images/packetscan_img36.gif")){kind=link}
