Patent Publication Number: US-2011058671-A1

Title: Secure protocol terminal adapter

Description:
CROSS REFERENCE TO RELATED APPLICATION 
     This application claims the benefit of U.S. provisional application No. 61/240,516 filed Sep. 8, 2009, which is incorporated by reference as if fully set forth. 
    
    
     FIELD OF INVENTION 
     This invention relates to the field of telephony-based communication systems. 
     BACKGROUND 
     Voice communications networks may include two major networking technologies, circuit switched technology and Internet Protocol (IP) technology. Circuit switched networks use switches to establish a dedicated path, or circuit, for voice or data to be communicated. IP networks use packets of information that are individually addressed and routed to communicate data. That data may represent a secure voice call, for example. 
     In secure voice/data communications, a secure telephone, such as a Secure Terminal Equipment (STE) for example, may plug into a standard telephone wall jack and communicate either in an unencrypted or an encrypted (secure) mode via the circuit switched network with other devices on the circuit switched network. Because the STE uses circuit switching technology, it cannot directly communicate with devices on IP networks. 
     A gateway interconnects the circuit switched network and the IP network for purposes of secure communication. To use a gateway, however, the STE must connect to the gateway via the circuit switched network. 
     SUMMARY 
     The secure protocol terminal adapter, disclosed herein, enables telephony-based, circuit switched endpoints to directly communicate with devices on the IP network. For example, a communication system may include an analog secure telephony terminal, a remote secure telephony terminal configured to communicate with an Internet Protocol network, and a terminal adapter. The analog secure telephony terminal may be, for example, a Secure Terminal Equipment (STE) device, an OMNI, a Sectera Wireline Terminal (SWT), or any other Secure Communications Interoperability Protocol (SCIP) enabled device. The terminal adapter may enable Internet Protocol connectivity for the analog secure telephony terminal. The analog secure telephony terminal may be connected directly to a terminal adapter. The terminal adapter may be configured to interface between the analog secure telephony terminal and the Internet Protocol network such that a secure channel is established between the remote secure terminal and the analog secure telephony terminal. 
     The terminal adapter may include a telephony station interface (such as a Foreign eXchange Station (FXS) interface, for example), a data communications interface (such as an Ethernet interface, for example), and a processing unit. The processing unit may be configured to establish a first connection (such as an Assured Services Session Initiation Protocol (AS-SIP) connection and/or a V.150 protocol connection, for example) over the data communications interface and a second connection (such as a V-series MODEM connection, for example) over the telephony station interface. The processing unit may be configured to communicate secure information (such as Type 1 encrypted voice and data information, for example) between the first connection and the second connection. The processing unit may include a V-series modem, V.150 stack and/or an AS-SIP stack. 
    
    
     
       BRIEF DESCRIPTION OF THE DRAWINGS 
         FIG. 1  is a block diagram of an example communications network. 
         FIG. 2  is a block diagram of an example secure voice/data protocol terminal adapter. 
         FIG. 3  is a flow diagram of an example method of protocol interworking for secure voice/data communication. 
         FIG. 4  is a sequence diagram of an example method of protocol interworking for secure voice/data communication. 
     
    
    
     DETAILED DESCRIPTION 
     An example secure voice/data protocol terminal adapter may include an analog telephone interface, a first converter stage, a second dynamic converter stage, an IP telephony protocol stack, an Ethernet Media Access Controller (MAC), and an interface to the Ethernet network. In operation, the analog telephone interface may receive analog voice signals from an analog telephone or analog data signals from an unencrypted data terminal, encrypted voice terminal, or encrypted data terminal. The first converter stage converts analog signals to digital signals. The second dynamic converter stage provides data compression and formatting dependent upon the type of signals being transmitted and includes an audio or voice CODEC, a MODEM, and a V.150 internetworking function. The analog or voice CODEC compresses and formats unencrypted audio. The MODEM modulates/demodulates unencrypted data, encrypted data, or encrypted voice communications. The V.150 internetworking function compresses and formats unencrypted data, encrypted data, or encrypted voice communications. The IP telephony protocol stack performs call management within the IP telephony infrastructure utilizing protocols such as Assured Service-Session Initiation Protocol (AS-SIP), Session Initiation Protocol (SIP), Real-time Transport Protocol (RTP), and Real-time Control Protocol (RTCP). The Ethernet MAC performs Ethernet interface connection management. The Ethernet network interface connects the device to the Ethernet network and IP telephony infrastructure. 
     Encrypted analog telephony users commonly utilize the Multilevel Precedence and Preemption (MLPP) capabilities within the analog telephony infrastructure. AS-SIP is implemented within the IP telephony protocol stack and performs MLPP within the IP telephony infrastructure. Unencrypted data, encrypted data, and encrypted voice analog telephony devices utilize modem technology from the International Telecommunication Union—Telecommunication Standardization Sector (ITU-T) V-series of Recommendations. The V.150 Modem Relay Protocol is implemented to support an internetworking function relaying the bi-directional V-series modem protocols over the IP telephony infrastructure in a reliable and bandwidth efficient manner. 
       FIG. 1  is a block diagram of an example communications network  100 . As illustrated in  FIG. 1 , the communications network may include an IP network  110  and a circuit switched network, e.g. the public switched telephone network (PSTN)  102 . Voice over IP (VoIP) devices such as a secure VoIP phone  120  and a VoIP phone  122  may communicate natively via the IP network  110 . However, traditional endpoints (such as computer modems  138 , analog phones  136 , and secure phones  134 ) may communicate via the IP network  110  either indirectly over the PSTN  102  and a V.150 gateway  104  or directly using a terminal adapter  132 , e.g. an AS-SIP Analog Terminal Adapter (ATA). 
       FIG. 2  illustrates an example secure voice/data protocol terminal adapter  200 . Such a terminal adapter may be used to connect legacy devices directly to an IP network  110 , as illustrated in  FIG. 1 . The terminal adapter may include telephony station interface  220 , a data communications interface  222 , and a processing unit  224 . The telephony station interface  220  may include an analog telephone connection  240 . The data communications interface  222  may include an Ethernet MAC  242  and/or Ethernet network connection  244 . The processing unit  224  may include an analog/digital-digital/analog (A/D-D/A) converter  230 , a CODEC  236 , a MODEM  232 , a V.150 internetworking function  204 , and/or an IP telephony protocol stack  206 . The elements and arrangement of the elements of the adapter and processing unit are illustrative and other elements or arrangements may be used. 
     The analog telephone interface may include one or more traditional RJ-11/RJ-14/RJ-25 6 position 2, 4, or 6 conductor sockets or jacks. This is the socket or jack found on traditional analog telephone handsets and analog telephone wall jacks. This connector type allows the adapter to interface with a variety of analog telephone equipment including but not limited to telephones, modems, secure telephone terminals, and fax machines. The analog telephone interface carries analog audio to and from the adapter that may be in the form of voice, modulated data or fax information, or dial/notification tones. For example, the analog telephone interface may provide what is referred to in telephony infrastructures as a Foreign eXchange Station (FXS) interface. For example, the analog telephone interface provides voltage to the connected analog telephone equipment as well as a dial tone and analog audio. The analog interface receives any analog audio sent from the attached analog telephone equipment. 
     The analog telephone connection  240  may be connected to an A/D-D/A converter  230 . The A/D-D/A converter  230  provides analog to digital and the reverse digital to analog conversion. The A/D-D/A converter  230  may be implemented within an Applications-Specific Integrated Circuit (ASIC) or a mixed signals System on a Chip (SoC) processor, for example. The A/D-D/A converter  230  may be coupled with the analog telephone interface such that the converter can digitize incoming audio from the analog telephone interface and pass the digitized audio on to either the CODEC  236  or MODEM  232  functions. The A/D-D/A converter  230  may also be coupled with the CODEC  236  or MODEM  232  functions such that any digitized audio from the CODEC  236  or MODEM  232  may be represented on the analog telephone interface as analog audio. 
     The CODEC  236  performs compression and decompression of digitized audio signals. The CODEC  236  functionality may be performed by an ASIC or Digital Signal Processor (DSP). The CODEC  236  functionality may perform, but is not limited to, one or more of the following audio compression algorithms: G.711 μ-law, G.711 A-law, G.718, G.719, G.722, G.722.1, G.722.2, G.723, G.723.1, G.726, G.728, G.729, G.729.1, Global System for Mobile Communication (GSM) CODEC, Speex, Vorbis, or Internet Low Bitrate CODEC (iLBC). The type of algorithm may be configured in the adapter. The adapter may also negotiate with a communicating party which compression algorithm should be used within the CODEC  236 . Configuration and/or negotiation of the compression algorithm may be accomplished by the IP telephony protocol stack  206 . 
     The CODEC  236  receives uncompressed digital audio signals from the A/D-D/A converter  230 , compresses the signals using the chosen algorithm, and delivers the compressed digitized audio to the IP telephony stack  206 . In the reverse direction, the CODEC  236  receives compressed digitized audio from the IP telephony stack  206 , decompresses the compressed digitized audio using the chosen algorithm, and delivers the decompressed digitized audio to the A/D-D/A converter  230 . 
     The MODEM  232  performs the modulation and demodulation functionality as described in the ITU-T V-series Recommendations on data communications over the telephone network. The MODEM  232  functionality can be performed within an ASIC, DSP, and/or general-purpose processor. The MODEM  232  may perform, but is not limited to, one or more of the following V-series ITU-T Recommendations on data communications over the telephone network: V.8, V.21, V.22, V.22bis, V.23, V.24, V.27ter, V.28, V.29, V.32, V.32bis, V.33, V.34, V.34bis, V.41, V.42, V.42bis, V.44, V.90, and/or V.92. The MODEM  232  is used to encode and decode data that is passed between the invention and a far end modem device performing encoding and decoding that is connected to the analog telephone interface. 
     The MODEM  232  receives uncompressed digital audio signals which carry modulated or encoded data from the far end modem via the A/D-D/A converter  230 , performs one or more features of ITU-T V-series Recommendations to demodulate or decode the data, and then delivers the decoded data to the V.150 internetworking function  204 . In the reverse direction the MODEM  232  receives data from the V.150 internetworking function  204 , encodes the data for transmission to the far side modem connected to the analog telephone interface using one or more of the ITU-T V-series Recommendations, and then passes the encoded data to the A/D-D/A converter  230 . 
     The V.150 internetworking function  204  performs portions of, but is not limited to, the ITU-T V.150 and V.150.1 Recommendations for Modem-over-IP networks, and their subsequent revisions. The V.150 and V.150.1 Recommendations detail how to establish, negotiate, transition between, maintain, and teardown Modem-over-IP connections. The Recommendations detail the use of IP control mechanisms and protocols as well as IP transport mechanisms and protocols to use to accomplish transparent end-to-end modem communications over an IP infrastructure. The V.150 and V.150.1 Recommendations reference the performance of V-series modulations; within the invention these modulations are performed within the MODEM  232 . The V.150 internetworking function  204  can be performed within an ASIC, DSP, and/or general-purpose processor. The V.150 internetworking function  204  receives data from the MODEM  232  and performs the adaptation of the data for transmission over an IP network, the data is then passed to the IP telephony protocol stack  206 . In the reverse direction, the V.150 internetworking function  204  receives packetized data from the IP telephony protocol stack  206  and adapts the data for transmission over a V-series modem, the data is then passed to the MODEM  232 . 
     The IP telephony protocol stack  206  performs the call management functionality within the IP telephony infrastructure for the invention. This capability can be performed within a general-purpose processor. The primary function of the IP telephony protocol stack  206  is to perform the AS-SIP call management protocol and a suite of supporting IP telephony protocols that may include, but is not limited to: IP, IPv6, ARP, TCP, UDP, DHCP, HTTP, TFTP, FTP, SFTP, SSH, SMTP, TLS, SSL, H.323, SDP, SIP, SCCP, MGCP, SCTP, RTP, RTCP, SPRT, SRTP, SRTCP, UDP-TL, MIKEY, and ZRTP. 
     AS-SIP allows the user to perform classic telephony tasks, such as, but not limited to, call placement, but also permits, but is not limited to, the user to performing MLPP. MLPP permits a user to select from a striated higher level of service and, if necessary, preempt network capacity to ensure that the user&#39;s call is capable of being established and remaining active. 
     Responsibilities of the IP telephony protocol stack  206  include, but are not limited to, telephony management tasks such as managing network presence, call setup, call initialization, call preemption, call precedence management, parameter negotiation, parameter exchange, managing call state transitions between voice and data modes, call notifications, call forwarding, and call termination. The IP telephony protocol stack  206  is also responsible for managing device configuration. Device configuration may be accomplished using one of the previously mentioned support protocols such as, but not limited to, HTTP using a configuration web page. Device configuration may alternatively be performed utilizing, but not limited to, an automated mechanism such as configuration file retrieval from a TFTP server that is defined by a DHCP server response. 
     Call data from the CODEC  236  or V.150 internetworking function  204  may be processed by the IP telephony protocol stack  206  and then passed to the Ethernet MAC  242  as a complete IP packet. In the reverse direction, complete IP packets carrying call data may be delivered to the IP telephony protocol stack  206  from the Ethernet MAC  242  and the encapsulated call data is passed to either the CODEC  236  or V.150 internetworking function  204  depending upon the mode of the call. When the call is in voice mode, call data may be passed to and from the CODEC  236  and when the call is a data or modem relay call, the call data may be passed to and from the V.150 internetworking function  204 . 
     The Ethernet MAC  242  manages the inventions participation in the layer 2 Ethernet network. The capability may be performed in a dedicated ASIC or general-purpose processor. The Ethernet MAC  242  may handle Ethernet datagram delivery and reception between the IP telephony protocol stack  206  and other Ethernet entities present on the Ethernet network segment  244 . 
     The Ethernet interface may consist of one or more traditional RJ-45 8 position 8 conductor sockets or jacks. This is the socket or jack found on traditional 10 BASE T, 100 BASE T and 1000 BASE T Ethernet devices. This connector type allows the invention to interface with a variety of Ethernet equipment including but not limited to hubs, switches, routers, computers, access points, and Ethernet gateway devices. Implementation of the adapter may possess one or more connectors representing one or many Ethernet network connections. The Ethernet network interface carries Ethernet datagrams back and forth between the Ethernet MAC  242  and the Ethernet network segment  244 . 
       FIG. 3  is a flow diagram of an example method of protocol interworking for secure voice/data communication  300 . The process  300  shown in  FIG. 3  may be implemented using the terminal adapter disclosed in  FIG. 2 , for example. The process incorporates MLPP processing once the user goes off hook  305 . A call is set up using the analog/voice CODEC  330 . If modem tones are detected in the audio  340 , then the call may transition to a V.150 interworking call  355 . If the modem tones are no longer detected in the audio, the call may transition back to an analog/voice CODEC call  335 . 
     First, a user goes off hook  305 . A determination is made whether a user requires MLPP  310 . If the user requires MLPP, then the user requests MLPP using the phone keypad  315  and the user dials a phone number  320 . If the user does not require MLPP, then the user can immediately dial a phone number  320 . A gateway device performs call setup with IP infrastructure  325 , a call is established using analog/voice CODEC  330  and participation is enabled in an analog/voice CODEC call  335 . If modem tones are detected in the audio  340 , the call transitions to V.150  335  and participation is enabled in the V.150 call  350 . If no modem tones are detected in the audio  345 , the call transitions back to participation in an analog/voice CODEC call  335 . The call can be terminated  360  after the call transitions to an analog/voice CODEC call  335 , or after the call transitions to a V.150 call  350 . 
       FIG. 4  is a sequence diagram of an example method of protocol interworking for secure voice/data communication  400 . In  FIG. 4 , a secure phone  402  communicates using a terminal adapter  404  via the IP network to a gateway  406  to the PSTN  408  and a traditional analog secure phone  410 . A similar call flow may be used to set up a call between the secure phone  402  with terminal adapter  404  and a VoIP secure phone, in which the gateway  406  and PSTN  408  signaling is substituted with the terminal signaling to a VoIP secure phone. 
     A user selects the desired MLPP level of the call  415  at the secure phone  402 . An MLPP level is selected  416  between the secure phone  402  and the terminal adapter  404 . The user dials a phone number  420  and after dialing 421, an analog/voice mode is selected  425  at a terminal adapter  404 . The terminal adapter  404  and gateway  406  complete AS-SIP call setup with selected precedence and number  426 . An AS-SIP link is established  430  between the terminal adapter  404  and the gateway  406 . Between the gateway  406  and the PSTN  408 , a PSTN call setup is performed with selected precedence and number  431 . A PSTN link is established  435  between the gateway  406  and the PSTN  408 . Ring tones and rings are exchanged  436  between a secure phone  402 , a terminal adapter  404  via the IP network, a gateway  406 , the PSTN  408  and a traditional secure phone  410 . The user picks up the call  440  at the secure phone  410  and end-to-end voice call is established  445 . A user at the secure phone  402  can request to “Go Secure”  450 . This initiates modem training  451  and at the terminal adapter  404  a switch to V.150 modem bypass mode occurs  455 . This completes modem training  456  between the secure phone  402  and the terminal adapter  404 . A modem link is established  460  between the secure phone  402  and the terminal adapter  404 . The terminal adapter  404  initiates V.150 connection setup  461  with the gateway  406 . The gateway  406  initiates modem training  462  with the secure phone  410  which completes modem training  463  with the gateway  406 . A modem link is established  465  between the gateway  406  and the secure phone  410 . This completes V.150 connection setup  466  between the gateway  406  and the terminal adapter  404 . A V.150 link is established between the gateway  406  and the terminal adapter  404 . Finally, a secure end-to-end call is established  475  between the secure phone  402  and the other secure phone  410 . 
     Features and elements are described above in particular combinations, each feature or element can be used alone without the other features and elements or in various combinations with or without other features and elements.