IP/VoIP Analysis & Simulation

Protocol Analyzer

Testing ED-137 and ED-138 Interoperability Standards for VoIP Air Traffic Control

Critical Delay Measurement, Voice Quality Testing & ED-138 Monitoring System

  ATM Test Solutions   MAPS™ ED-137 Radio

  MAPS™ ED-137 Telephony   MAPS™ ED-137 Recorder
Air Traffic Management  - ED-137 & ED-138 Standards for CoIP Air Traffic  Conttol

Overview

Voice communications for air traffic management, whether it is Air-Ground (A-G) or Ground-Ground (G-G) were predominantly over TDM based Air Traffic Management (ATM) networks. With the latest developments in EUROCAE (European Organization for Civil Aviation Equipment) ED-137 inter-operability standard, it is now possible to implement VoIP technology for voice services for air traffic control. This implementation uses the familiar Session Initiation Protocol (SIP) to establish, modify, and terminate sessions within an Air Traffic Services Ground Voice Network with endpoint equipment. The endpoint equipment can be SIP based Controller Working Positions (CWP), Next Generation Voice Communication Systems (VCS), Radios, Recorders, and VCS/Radio Gateways allowing interoperability with legacy equipment and protocols as shown in the diagram above. Existing VCS can access an IP wide area network as the connection backbone using gateways.

Before such a system is made operational, it must be thoroughly tested for voice quality, latency, performance, reliability, and functionality. GL tools offer end-to-end testing from the radio interfaces to the CWP.

GL offers the following solutions for ATM networks:

Product Videos

Critical Delay Measurements using Audio Analyzer & Packet Analyzer

Measuring delay, jitter and packet loss through air traffic networks is crucial and network testers must precisely time events. Delay testing and verification will be particularly important with the transition to IP networks, which will introduce packetization delays, jitter buffer delays and other uncertainties. For example, measuring the time from which the Air Traffic Controller presses ‘Push To Talk’ to the time when the IP packet indicates that the PTT bit is set to 1 is an essential testing scenario. This delay measurement is possible using GL’s Audio Analyzer as well as GL’s Packet Analyzer.

Testing for voice quality and one-way delay of the Analog / TDM and VoIP systems can be done using GL's Audio Analyzer by transmitting and recording analog signals at the CWP, Radio and VoIP gateway interfaces. The Audio Analyzer can automatically key PTT from the CWP and then transmit and receive audio for precise delay measurements. The Audio Analyzer can generate triggers based on PTT activation. Below is a list of the key features associated with the Audio Analyzer:


  • 4-Wire Balanced Audio and PTT Contact Closure Interface for connecting to the CWP Dual Jack Module as well as other 4-wire interfaces with in the network
  • Precise one-way delay measurements and event-based TTL or CMOS trigger outputs that can feed into an oscilloscope
  • Voice Quality Assessment using industry-standard ITU algorithms (PESQ &POLQA)
  • 4-Wire “Bridge” Mode – To monitor existing TMG’s and provide event-based trigger outputs (for latency testing)
  • Test Automation Capabilities - GUI-based or SDK and CLI versions

The Packet Analyzer can generate triggers based on packet filters. The Packet Analyzer also includes a wirespeed filter that allows a user to filter out unwanted traffic and continuously capture the traffic of interest. Features of the wirespeed packet filter are listed below:

  • Filter packets and record only packets of interest
  • Capture simultaneously on 2 ports with 40 bytes deep filter per port (for record Only module) or on 3 ports with 16 bytes deep filter (for Record and Playback module) and set filter on any one of the ports or all ports
  • Packet filtering can be based on layer 2 (Ethernet), layer 3 (IP) or layer 4 (UDP/TCP) headers
  • Generates a trigger (1 Microsecond pulse) for each packet that satisfies filter criteria

For more details, please visit Critical Delay Measurements in Air Traffic Management


The depiction includes various GL test tools such as the Audio Analyzer, Packet Analyzer, TTL Signal Packetizer, and Event Data Logger. All these components are controlled by a centralized component called MAPS™ Administrator. MAPS™ Administrator calculates the time difference between posted events, i.e., Discrete Events (from the Packetizer) and Timed Events (from Packet Analyzer) and reports precise measured delay at different points in the network. All the components support a client server model, with the MAPS™ Administrator acting as a client and controlling all the other components which act as servers.

GL’s TM-ATM Solution for Timing Measurements


Refer to Delay Measurements section for detailed information about the GL Test Tools used in the solution.

ED-137 Test Solutions

GL’s MAPS™ ED-137 address the need for additional testing needs that may include simulating many endpoints and generating bulk calls (load testing) on the network. GL offers (Message Automation & Protocol Simulation) MAPS™, a software platform, which can simulate both Air-Ground calls (as per ED137_1B and ED137_1C: Radio) and Ground-Ground (as per ED137_2B and ED137_2C: Telephone) calls as per EUROCAE (European Organization for Civil Aviation Equipment) ED-137 standards. MAPS™ ED-137 support can emulate many user agents (end points) without requiring analog interfaces.

The software also supports simulation of Recorder interface for both Air-to-Ground and Ground-to-Ground calls at CWP, GRS and Recorder endpoints as per ED137_4B and ED137_4C versions. Air Traffic Recorder is the next generation VoIP recorder implemented as per ED-137 inter-operability standards. Specially designed for all traffic control towers and centers to simplify the recording, archiving, and playback voice communications. It can simulate AG/GG call recording towards Recorder and testing Recorder interface of CWP/VCS and GRS in ATM network.

The following simulators are ED-137 B and C versions compliant and are VOTER validated:

MAPS™ ED-137 VoIP ATM Telephone can simulate hundreds of Ground-to-Ground calls, supporting all Telephone call types and scenarios such as Call Hold and Call Transfer.

MAPS™ ED-137 VoIP ATM Radio can be configured as CWP and GRS to simulate outgoing and incoming messages in Air-to-Ground call.

MAPS™ ED-137 Air Traffic Recorder can emulate call recording functionality at CWP, GRS and Recorder interfaces, generating more than hundreds of recording sessions to verify performance and load testing.


MAPS™ ED137 can be used to set up voice sessions over a network, then send and record test voice signals for assessing voice quality and performance. It simulates both Controller Working Position and Radio Media Gateway System, with following important features required to maintain reliable communication over air traffic network.

  • RTP Extended Header
    Supports RTP header extension format as per extended header type 0x167 (WG67). Both audio RTP and R2S packet will carry this 8 byte extended header. PTT and Squelch fields are set properly to signal voice activity in uplink or downlink. PTT-id assigned by remote end will be reflected in extended header.
  • R2S-KeepAlive packets
    Supports generation of R2S-KeepAlive packets while there is no active voice being transmitted or received to keep the SIP session maintained. Push-To-Talk (PTT) and Squelch fields are reset properly to signal silence (idle period) in uplink and downlink respectively.
  • Supports additional SDP parameters
    • R2S-Keep Alive Period: Maximum time (or frame interval) between each R2S-keep Alive packet. integer, in milliseconds.
    • R2S-Keep Alive Multiplier: Number of Keep Alive packets in error or not received by endpoint before Time Out of the session
    • PTT ID: PTT identity (value from 000001 to max. number of possible RTP streams, e.g.111111=63) assigned to each user agent by GRS endpoint in 200OK response to INVITE.
    • Version of the used RTP HE (Header Extension).
    • Frequency Identity.
  • Supports Additional SIP headers 'Priority' and 'Subject’.
    Every INVITE contains these two headers.
    • Priority: normal (Indicates SIP call priority)
    • Subject: radio (Indicates SIP call type)
  • IP Spoofing
    Supports IP address spoofing to generate each SIP call using different IP address from a single system.

For more details on MAPS™ ED-137, please visit MAPS™ ED-137 Telephone , MAPS™ ED-137 Radio, MAPS™ ED-137 Recorder webpages.

WAN IP Link Simulation using GL’s IPNetSim™

GL can emulate various impairments that occur on Wide Area Networks (WAN) using a tool called IPNetSim™. This appliance is connected in-line on an Ethernet network and acts as a transparent bidirectional Ethernet link. The different WAN link speeds that can be emulated are - RS232/DSL/Modem/T1 E1/T3 E3 and more.

IPNetSim™ can simulate the impairments listed below:

  • Bandwidth emulation from 100 Kbps up to 10 Gbps in increments of 1 Kbps emulating various WAN technologies like Modem, DSL, T1/E1/T3/E3/OC3/OC12 etc
  • Latency/Delay emulation from 0 milliseconds to 8 seconds, with single, uniform and random distributed delay emulation capabilities
  • Packet Loss emulation, from 0 to 100%
  • Packet Reordering emulation, from 0 to 100% with random reinsertion delay
  • Packet Duplication emulation, as a percentage of total packets, from 0 to 100%
  • Logic Error Insertion - inserts error anywhere with in the frame, from 10-1 to 10-9 rate

Voice Quality Testing (VQT)

The figure above shows MAPS™ ED137 simulating a call from an aircraft, by sending a voice file over the radio network to an air traffic controller. When there is no voice transmission from the aircraft, only keep-alive packets are sent from radio towards the VCS/CWP endpoint. The PTT keying by a pilot triggers voice transmission towards the ground and keep-alive packets are replaced by audio packets from radio towards the VCS/CWP. The voice traffic is received at the controller side and recorded. With GL’s Voice Quality Testing application, the received file is compared against the original and a standard ITU-based voice quality score is derived. The RTP traffic can also be converted to analog and output to a PC speaker.

For more details on GL’s VQT, please visit Voice Quality Testing - PESQ-POLQA

Testing Radio/Network Links – Answer and Send Voice with Squelch