Call and Traffic Generation/Reception |
Results and Displays
Configuration Parameters |
Call and Traffic Generation/Reception
Manual Call Generation
Included with the DCOSS Manual Call Generation user interface, the user may manually place a call,
answer a call, reject a call, or hang-up a call, on a specific timeslot, without the telephone handset. This allows great flexibility with
setup and release of a call.
Screenshot of Manual Call Generation
Bulk Call Generation/Reception
Using the DCOSS graphical user interface, automatic bulk calls generation and reception is possible for one, many or all timeslots of the T1 and E1
trunks. This allows load testing of the digital trunks.
Automated call generation allows the user to automatically generate a multitude of calls, for one, many, or all specified timeslots and
dictate the calling process of each call individually.
Along with user-defined parameters, each timeslot may be configured with the following properties:
- Send single or multiple user specified PCM voice file(s) (a current file located on computer hard disk)
- Record to a user specified PCM voice file and perform Voice Quality Testing
- Transmit a user specified FAX file
- Send user specified DTMF or MF digits (The DTMF/MF digits may include randomized values, as well as incremental values. This provides greater
flexibility during the testing process.)
- Send user specified Single- or Multi- Frequency Tones
- Establish a Modem Traffic Call
Screenshot of Bulk Call Generation/Reception
Bulk Call Scripting
Bulk Call Scripting allows the user to custom configure all aspects of each time slot during bulk calling. This includes, but is not limited to,
sending PCM Files as well as DTMF digits, responding to detected digits or frequency tones, and performing randomized events per call. The Bulk Call
Scripting function utilizes a point-and-click user interface for creating each individual script. Each script may be used several times for
different timeslots. The user will have an option to either setup bulk calling, for an individual timeslot, using the standard configuration settings or
to use a pre-configured bulk call script.
Bulk call scripting also includes answering incoming calls with auto busy option, use of wildcards, and rejecting incoming calls based on
timeslot, incoming phone number, user specified cause (ISDN/SS7 only), or SIT tone.
Screenshot of Bulk Call Scripting
Screenshot of Bulk Call Scripting Editor
CAS State Machine
DCOSS CAS State Machine is an optional application that simulates any user-defined CAS protocol by providing signaling bit transitions and
forward/backward frequency tones/digits. The supported protocols include - E1 MFC-R2 (All variants, full / semi compelled), T1 Winkstart (R1 wink),
T1 Loopstart and T1 Groundstart, E1 European Digital CAS (EUC), and anu user-defined CAS protocol.
The GLís DCOSS CAS State Machine provides a script editor, which is based on a self-descriptive language that can define the behavior of
CAS Call Control procedure. Functions such as Place Call, Answer Call, Incoming Call, and Disconnect Call are defined within the script. Additionally,
more advanced script may also be defined in the script editor. User may define Signaling Bit Transitions and forward/backward digits/tones within each script.
Screenshot of CAS State Machine
Screenshot of CAS State Machine Script Editor
PCM Voice File Generation/Reception
Using the DCOSS, after a call is established a PCM voice file(s) may be transmitted (in 8-bit A-law & µ-law, 16-bit raw & WAV file formats)
transmitted over the user specified timeslots of the E1 and T1 trunks. This allows confirmation of voice quality over the established call.
Single or multiple PCM or WAV voice files may be transmitted over a single timeslot or over several timeslots.
In addition, the DCOSS Send PCM Voice Files may be placed in Continuous Mode. While in Continuous Mode, all PCM or WAV Voice
Files being transmitted will continuously transmit until interrupted, by the user, via the DCOSS user interface.
The DCOSS Record PCM Voice File will record, to a user specified PCM (as either A-Law or µ-Law) or WAV Voice File, all communication over a user
specified timeslot after a call is established over the same timeslot. Recording to a PCM or WAV Voice File is possible over a single open timeslot, as
well as recording from multiple timeslots to multiple PCM or WAV Voice Files. Several user-defined parameters are available while recording a
PCM or WAV Voice File.
The user may configure several parameters associated with sending/recording PCM or WAV voice files. These parameters include tx/rx gain and specific
WAV file parameters.
Using the DCOSS Send PCM and Record PCM features, the user can perform Voice Quality Testing (VQT) using POLQA and PESQ
analytic measurement tools. Please refer to the VQT section for more detailed information.
Screenshot of Send/Record PCM Voice Files
Using the DCOSS, the user may generate a real time FAX using a user-specified file (TIF format).
The Send FAX function may be configured to either send a FAX immediately after a call is established, or manually send a FAX during an
established call. The Receive FAX function may be configured to either automatically answer a call and begin receiving a FAX (after detection of the
CNG tone) or manually receive a FAX during an established call. The resultant fax will be placed in a user-specified file.
DCOSS FAX may also be configured for continuous mode for sending/receiving manual faxes. The user may also globally start
sending/receiving manual faxes over all specified timeslots/trunks at the same time.
During the Send FAX and/or Receive FAX function, DCOSS will perform all required handshaking to establish and generate/receive the user
specified FAX file. DCOSS may be configured to send/receive faxes over all timeslots of all trunks simultaneously.
The DCOSS FAX supports V.29, V.27, V.33, and V.17 fax protocols for both transmit and receive.
Within the FAX function, the user can specify Fax Quality Test (FQT) parameters. Using FQT the user can assure that the specified fax was sent
and received without errors or deficiencies.
Screenshot of Fax Generation/Reception
Using the DCOSS, the user may generate modem traffic and send/receive text files once the modems have properly connected.
The modem function may be configured to establish the modem connection immediately after a call is established or manually during an
established call. Once the modem call is established, a file or plain text can be sent and received. Modem Quality Testing (MQT) is also
associated with the modem call. This includes testing for modem speed, proper training, and a bit-by-bit comparison between the sent file
and the received file.
The DCOSS modem supports V.21, V.23, V.34, V.90, and V.92 modem protocols.
Screenshot of Modem Generation/Reception
DTMF/MF Digit Generation/Reception
Using the DCOSS, after a call is established the user may specify DTMF or MF digits to transmit over the E1 or T1 digital trunk(s).
The user may specify the digits to send individually for each of the timeslots using the DCOSS DTMF/MF Digit Generation user interface.
The user may also send DTMF or MF digits to all timeslots simultaneously using the digit keypad.
Detection of all DTMF/MF digits is displayed, per open timeslot, and may be recorded to a flat text file.
Screenshot of DTMF/MF Digit Generation/Reception
Frequency Tone Generation/Reception
Using the DCOSS, after a call is established the user may specify a Single-Frequency or Multi- Frequency Tone to transmit over the E1 or T1 trunk(s).
The user may specify the frequency, amplitude, as well as the OnTime and OffTime for each generated Frequency Tone. The user may also specify
the frequency and bandwidth for detecting frequency tones. The detected frequency tones may be recorded to a ASCII file.
Screenshot of Frequency Tone Generation/Reception
Results and Displays
Call Record Generation
An important aspect of a Central Office Switch is the ability to log the call records, both complete as well as incomplete calls.
The DCOSS will display, to the main screen, all complete and incomplete calls.
Included with the Call Record Generation function, the call record data may be queried in order to display only
those call records requested. The call record data may be saved to an ASCII text file.
Screenshot of Record Generation
Call Quality Test Criteria
Associated with the Call Records, the user may specify criteria for Passing or Failing a completed call. The user-specified criteria will allow the user
to make certain that the digital voice/data channel was clear. If a call fails the user-specified criteria, an event will be generated on the DCOSS
Captured Error Event (Failed Call) screen. This event will display the timestamp/date, timeslot/trunk and which user-specified criteria(s) failed.
Screenshot of Call Quality Test Criteria
Real Time Call Status
Current Status of the trunk line along with all established calls is displayed
within the DCOSS Call Status screen. The Real Time Call Status gives an overview of all timeslots, currently active, for each of the T1 and E1
Screenshot of DCOSS Timeslot Status
The DCOSS Call Statistics screen displays, in real-time, statistics for all timeslots and all trunks. These statistics include the following: Call
Attempts, Incoming Calls, Call Complete, Busy Calls, Incomplete Calls, Errors, DTMF Digit Detection, Tone Detection, PCM File Done, Fax Generated,
and Fax Completed. The statistics screen is updated in real-time and may be output to an ASCII file.
Screenshot of DCOSS Call Statistics
Within DCOSS there are many functions and features and there are many parameters/options associated with these functions/features.
The DCOSS System Status screen displays, at a glance, the current status of all options/parameters of the DCOSS. This is quite useful in determining
what is currently configured without having to display each of the many DCOSS Function/Feature screens.
Screenshot of DCOSS System Status
All events within DCOSS are captured to screens for viewing or sending to an ASCII file. Events captured include normal call protocol events
(i.e. Seizure Detected, Incoming Call, Answering Call,) as well as Error Events. Included with the SS7 protocol, all SS7 events are also captured
and displayed. Included with the SS7 events are all pertinent information associated with each SS7 message (message type, CIC, OPC, DPC).
Each of the captured event screens may be frozen (display last 100 events only), dumped to an ASCII file, or saved to an ASCII file in real time.
Also, a filtering mechanism is available in order to view only those messages/events defined by the user.
Screenshot of DCOSS Captured Events
Digital Trunk Alarms
DCOSS will display, in real time, loss of sync or loss of signal for all digital trunks. This information is also displayed on the DCOSS Main Menu for
immediate detection of a specific trunk loss.
Screenshot of Digital Trunk Alarm Status
Board Configuration Setup
The digital trunks may be configured, per trunk board, using the DCOSS user interface shown below.
DCOSS System Specific has Clock Reference options - Internal (OSC) or External (using a desired digital trunk from Board 0). All trunks will grab
the clock from the user-defined single clock reference
Screenshot of DCOSS Board Configuration Setup
Screenshot of DCOSS System Specific Setup
The DCOSS Energy Detector function allows detection of energy and/or silence of all timeslots of all trunks. The user may configure the
parameters required for energy/silence detection, as shown below.
Screenshot of DCOSS Energy Detector
Phone/SS7 TCAP Routing
Based on the number dialed (using an analog phone), the DCOSS will re-route the outgoing call to a user-specified timeslot and trunk.
If the user-specified timeslot/trunk is currently unavailable the call may either continue on the original timeslot/trunk or the call will discontinue
and return a busy tone. The user can setup routing hunt groups while configuring the timeslots. The user may also specify whether the DID
Phone Routing will execute on the entire dialed number, extension of the dialed number, or the prefix of the dialed number. A maximum of
240 DID Phone Routing instances could be configured.
Screenshot of DCOSS Phone/SS7 TCAP Routing
DCOSS call switching incorporates two modes of operation.
Based on the number dialed (called/calling number), the DCOSS will switch the incoming call to a user-specified timeslot/trunk.
Based on a user-specified timeslot/trunk, the DCOSS will switch the incoming call to a user-specified timeslot/trunk. Thus, if timeslot 1/trunk 0
is configured to switch to timeslot 5/trunk 0 and a call is detected on timeslot 1/trunk 0, the call will automatically be switched to timeslot 5/trunk 0,
regardless of the number dialed.
Switch routing allows DCOSS to act as a true CO switch and provide throughput over the entire call. If the user-specified timeslot/trunk is currently
unavailable the call may either continue on the original timeslot/trunk or the call will discontinue and return a busy tone. The user can setup
routing hunt groups while configuring the timeslots. The user may also specify whether the Switch Routing will execute on the entire dialed number,
extension of the dialed number, or the prefix of the dialed number. A maximum of 240 Switch Routing instances can be configured.
Screenshot of DCOSS Called # Switching
Screenshot of DCOSS Dedicated Timeslot Switching
DCOSS Analog Station Connect
Establishing calls via analog phones as well as confirming a voice path over an established call is a vital requirement for testing and analyzing
a digital trunk. The DCOSS system allows from 8 to 64 ports for analog station phones (available Station interface includes telephones, FAX Machines,
modems). Each phone is user-defined with either a four number telephone extension or a full phone number (up to 25 digits) used for real-time call
routing purposes. For each of the station phones, the DCOSS system will detect off-hook and on-hook and act accordingly.
Using the DCOSS Station Connect, a user may place a call (or receive a call) from one phone handset to another phone handset and confirm
voice path as well as voice quality over the open trunk channel. If a PCM Voice File or DTMF/MF digits are sent over the established line, voice quality
may be confirmed using the analog station phone.
As an added feature, each of the station phones may be independently configured for Monitor mode, each with a user-specified trunk/timeslot.
This allows confirmation of voice quality during Manual Call Generation as well as Bulk Call Generation. In addition, a user may monitor both
sides of an established call (using DCOSS for both the outbound and inbound sides of the established call).
Screenshot of DCOSS System Setup
Screenshot of Station Board Parameters
Screenshot of System Parameters
Screenshot of Protocol Related
DCOSS BRI ISDN ST/U-Interface
The DCOSS BRI ISDN ST/U-Interface provides the capability of connecting up to sixteen (ST-interface) or four (U-interface) ISDN terminals
(telephones) to the DCOSS. Each of the ISDN ports has two 64 kbps B-channels and one 16 kbps D-channel. ISDN layers 1, 2, and 3, as defined in Q.931,
are supported within the DCOSS BRI ISDN interface.
The user may place or receive a call, using an ISDN terminal. The DCOSS displays the Q.931 messages during call setup and call teardown in real time.
Additional features of the DCOSS BRI ISDN interface are as follows:
- Multiple calls per port
- Place calls on hold
- Multiple call appearances on a single terminal
When the DCOSS is configured with both BRI ISDN as well as Analog Phones, a call may be placed using the following scenarios:
- Analog Phone to Analog Phone
- BRI ISDN phone to BRI ISDN phone
- BRI ISDN phone to Analog Phone
- Analog phone to BRI ISDN
Screenshot of ISDN-BRI Setup
Screenshot of ISDN BRI Terminal Equipment Setup
DCOSS VoIP Solution
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
DCOSS T3 Soultions
GL introduces a complete T3 digital Bulk Call/Manual Call emulation system. Using four DCOSS systems, supplying all 56 DS1's, and two
T3 Mux's, a fully loaded T3 system can be configured. Two DCOSS units (one with 16 T1 spans, the other with 12 T1 spans with full control of
signaling, traffic generation, and traffic detection) are required for fully loading a DS3 line. Thus, four DCOSS units are required for simulating both
sides of the DS3 line. The DCOSS Remote Client can control all four DCOSS systems using a single GUI.
Traffic (real-time Voice Calls, Voice Files, Fax Calls, Modem Calls, DTMF/MF digits, and Single-/Dual-Frequency Tones) can be individually assigned
to each circuit through Bulk Calling Configurations as well as Bulk Calling Scripting. For more information click
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