The Assured Services Analog Telephone Adapter (AS-ATA)
Information Assurance Specialists (IAS), Inc.’s Assured Services - Analog Telephone Adapter (AS-ATA) addresses the many problems associated with implementing SCIP enabled analog Secure Telephone devices within VoIP networks. The AS-ATA allows analog wire-line SCIP terminals such as the L-3 STE, L-3 OMNI, and GD Sectera Wireleine Terminal to be used within SIP based VoIP networks. By providing users of analog secure telephony products the capability to interface to VoIP networks the need to procure new or additional secure VoIP terminals from the various vendors is all but eliminated. The AS-ATA implements the Session Initiation Protocol in accordance with the DISA defined Assured Services Session Initiation Protocol (AS-SIP) specification. The use of DISA’s Assured Services implementation of SIP allows VoIP devices to incorporate the DOD’s requirement for a Multilevel Precedence and Preemption capability in voice networks. Additionally, the AS-ATA implements the V.150 Modem over IP protocol. By leveraging the V.150 protocol, the AS-ATA minimizes the per call consumption of bandwidth on the VoIP network when Secure Voice or Data is in use. Typical ATA devices require in excess of 160 Kbps of aggregate bandwidth to support secure voice or secure data communications. However, use of the AS-ATA to communicate Secure Voice or Data reduces aggregate bandwidth consumption by almost 75% over typical ATA devices. Additionally, the AS-ATA enhances call robustness, call success rate, and voice quality by leveraging bandwidth efficient and packet loss tolerant protocols. Using the AS-ATA ensures that secure voice and data communication over IP sessions remain robust in high packet-loss or high latency networks such as tactical, satellite, or cellular networks.
Application The AS-ATA allows users of the National Security Agency’s (NSA) Secure Communications Interoperability Protocol (SCIP) enabled POTS secure telephony products including, but not limited too, the L-3 STE, L-3 OMNI, GD SWT, GD SWT BDI, and vIPer PSTN to interface and communicate circuit switched clear voice, secure voice, clear data and secure data over IP telephony networks as a method of extending, replacing, or accessing the PSTN. AS-SIP is implemented in support of DOD VoIP network requirements for Multilevel Precedence and Preemption. The V.150 Modem Relay Protocol is implemented to support relaying the bi-directional V.22bis and V.34 modem protocols used by the SCIP enabled secure telephony products over the IP telephony infrastructure in a reliable and bandwidth efficient manner. V.150 is currently available in various gateway devices from several vendors. The AS-SIP/V.150 enabled ATA permits SCIP based secure communications devices to migrate to an IP telephony infrastructure and negates the need for VoIP specific SCIP terminals. The AS-ATA is available in several unique form factors; each designed to seamlessly blend into daily use of the L-3 STE, L-3 OMNI, or GD Sectera Wireline Terminal. Configuration of the AS-ATA is simplified to aid in ease of deployment. The VoIP Network Administrator simply provisions the AS-ATA device within a CallManager™ or SIP server, connects the device to the VoIP network, connects the analog terminal to the device, and the user is ready for secure communications. No additional User or Administrator configuration necessary.
- 1 RJ 45 Local Area Network 10/100 Base T Ethernet
- 1 RJ 45 Wide Area Network 10/100 Base T Ethernet
- 1 RJ 11 Analog Telephone Interface (FXS)
- 1 Power Interface
- Internet Protocol v4 and v6 compatible
- Assured Services Session Initiation Protocol (AS-SIP)
- Session Initiation Protocol (SIP)
- V.150 Modem Relay Protocol (V.150.1)
- G.711 μ-law voice Codec
- Real-time Transport Protocol (RTP)
- Real-time Transport Control Protocol (RTCP)