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GL Enhances MAPS™ ED-137 Telephone Emulator for VoIP Air Traffic Management

Gaithersburg, Maryland, USA - August 17, 2022 - GL Communications Inc., a global leader in telecom test and measurement solutions, addressed the press regarding their voice communication solution MAPS™ ED-137 Telephone for VoIP Air Traffic Management.

MAPS™ ED-137 Telephone

Overview

“GL’s Message Automation and Protocol Simulation (MAPS™) ED-137 Telephone Emulator can emulate Ground-to-Ground calls as per EUROCAE ED-137B Volume 2 and ED-137C Volume 2 Telephone standards. The software provides complete control over call scenarios and the ability to customize network parameters for signaling and VoIP traffic,” said Vijay Kulkarni, CEO of GL Communications.

He further added, “The software can be installed on any Windows® PC and uses the PC’s network interface card to generate and receive ED-137 traffic. The software can generate more than 500 simultaneous calls, emulate hundreds of user agents and full functionality of the Controller Working Position (CWP) entities in Ground-to-Ground telephone calls”.

Recent Enhancements


  • Enhanced to support Python Client Application Programming Interfaces (APIs) to operate in Command Line Interface (CLI) mode
  • Support Meet Me Conference call feature
  • Provides option to insert SIP headers during run-time in CLI mode
  • Validated against VOTER_4.1.30.3 for VCS-Telephony-Interface-Test Cases

Key Features


  • Portable and easy to use during field installation, system configuration, and commissioning
  • Supported call types include
    • Instantaneous Access
    • Priority Direct/Indirect Access
    • Routine Tactical Direct/Indirect Access
    • Routine Strategic Direct/Indirect Access
    • Routine General Purpose Direct/Indirect Access
    • Position Monitoring (Combined A-G and G-G, A-G only, and G-G only) calls
    • Addendum 2: FAA Legacy Telephone Interworking
    • Addendum 4: Override Call
    • Addendum 5: Voice Call
  • Emulate different call scenarios like Call Hold, Call Transfer (attended and unattended), Call Pick-up, Preset Conference call, Broadcast Conference call, etc.
  • Easy to understand Call Flow Graphs of SIP message exchanges and displays message contents (SIP headers and SDP attributes)
  • Allow call rejection using SIP error codes (4xx, 5xx, 6xx)
  • RTP Traffic can be impaired by packet loss, latency, duplicated and out-of-sequence packets
  • Allows the user to define DSCP (Differentiated Service Code Point) values for signaling and voice traffic
  • Support complete customization of SDP and SIP headers, call flow, and messages
  • Tests can be run sequentially, randomly, or simultaneously for multiple iterations
  • Run sets of test cases automatically at a predefined time with a Scheduler feature
  • Support IP address spoofing for each endpoint to generate calls using a unique IP address from a single system
  • Support audio codecs such as G711 U-Law, A-Law, and G729

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