Abstract:
A digital telecommunications system, a method of managing communications in such a system and a program product for managing audio transmission in a digital communications system can include devices at network endpoints selectively, transparently providing voice samples of sufficient quality for authentication and identification during conversations with the devices. The devices can respond to an authentication request by collecting authentication samples of an ongoing conversation with the samples having sufficient detail for authentication. The devices send the authentication samples in parallel that do not disrupt the conversation. Authentication samples may be verified prior to authentication by comparison against the corresponding portion of the ongoing conversation.

Description:
CROSS-REFERENCE TO RELATED APPLICATIONS 
     This application is the United States national phase under 35 U.S.C. §371 of PCT International Patent Application No. PCT/US2008/010731, filed on Sep. 15, 2008. This application is incorporated by reference herein. 
     BACKGROUND OF THE INVENTION 
     1. Field of the Invention 
     The present invention is related to voice identification and authentication systems and more particularly, to providing reliable voice identification and authentication in Voice over Internet Protocol (VoIP) based telecommunications systems. 
     2. Background Description 
     State of the art telecommunication systems are digital and, frequently, use Internet Protocol (IP) based communications. Unlike analog voice channels with a continuous analog signal, an IP communications system segments audio data, encodes and packetizes the segments and transmits the encoded IP packets between network entities in a connectionless transfer. Bearing in mind that the human ear has a range of no more than 20 Hertz (20 Hz)-20 KHz and typical telecommunications channels may have only bandwidth of hundreds of KHz, audio occupies a very small portion of a typical IP communication. Standards have been developed and promulgated for Voice over IP (VoIP) communications to insure that typical IP networks compensate for transmission delays and address Quality of Service (QoS) issues. These standards select small size for audio segments for encoding as relatively small packets and select transmitting those encoded small packets at a relatively high frequency such that decoding and transmission delays are unnoticeable or, at least, tolerable. 
     For example, G729 is one such standard audio data compression algorithm for VoIP, wherein raw audio is segmented, typically, into 10 millisecond segments and each segment is compressed in an IP packet. RFC 3551 defines a net audio data stream for a G729 code/decode (codec) with an 8-kbit/sec data rate. See, e.g., www.apps.ietf.org/rfc/rfc3551.html#sec-4.2. While the popular Gxxx telecommunications codecs, such as G723 or G729, provide for efficient package based voice communications, they may not provide adequate or even necessary support for high quality voice data required by state of the art voice recognition. 
     A growing number of various applications use voice recognition for voice authentication. Typically, these voice authenticated systems store voice signatures, e.g., in a database, that are used to authenticate a caller. These systems may use voice identification and authentication to grant access to sensitive personal data, such as identifying and authenticating bank customers for remote banking. Once authenticated, customers may be granted access respective bank accounts for remote home control with banking systems responding, e.g., using voice commands. Protecting such sensitive personal data and resources against unauthorized access is important to protect the respective customer&#39;s property. Other state of the art applications of voice recognition include, for example, using high quality voice signatures for lawful voice signed agreements and voice recorded contracts. These voice identification and authentication applications require high quality voice data for reliable identification and authentication at a quality not provided by standard telecommunications codecs. While traditional digital voice telecommunications codecs, such as G711 for example, or media based codecs (e.g., for music or video, such as MPEG) may transfer voice with high quality, sufficient quality to meet the authentication needs, VoIP telephony do not. 
     As noted hereinabove, the voice and audio in VoIP telephony are usually encoded and compressed to allow more efficient bandwidth usage. As further noted this encoding and compression may still allow suitable conversational voice content, it only needs to be sufficient for a human at one end of a conversation to use any of many voice features to recognize his/her partner in a communication. These voice features may include, for example, the partner&#39;s language, grammar, sentence building, tones, accents and/or voice patterns. However, a machine uses mainly sound related fewer features to recognize a speaker&#39;s voice. These features may include tones, accents and voice patterns that may not be included or encompassed by the popular telecommunications codecs. Thus, the audio data provided in normal telecommunications conversations is of insufficient quality for voice recognition, which is required for reliable identification, authentication and signatures. On the other hand, authenticating using a high quality compact disk (CD) encoding or other media codecs, e.g., sending only the authentication data in a MPEG derivative (e.g., mp3) fails to provide much security, if any. Further, using high quality communications (i.e., sufficient for transferring reliable identification, authentication and signatures) has typically proven to be too costly and to use far too much bandwidth and channel resources. 
     Thus, there is a need for satisfying the limits of narrowband voice communication systems, such as in state of the art VoIP telephony systems using high-compression codec for conversations, while enabling voice identification, voice authentication and voice signature communications to systems and applications that require high quality voice data. 
     SUMMARY OF THE INVENTION 
     It is a purpose of the invention to allow transferring real time voice identification, voice authentication and voice signature date in narrowband communications; 
     It is another purpose of the invention to facilitate transferring voice identification, voice authentication and voice signature transparently in VoIP communications in real time; 
     It is yet another purpose of the invention to allow transferring voice identification, voice authentication and voice signature transparently in real time during VoIP communications. 
     The present invention relates to a digital telecommunications system, a method of managing communications in such a system and a program product for managing audio transmission in a digital communications system. Devices at network endpoints, e.g., session initiation protocol (SIP) devices, selectively, transparently provide voice samples of sufficient quality for authentication and identification during conversations with the devices. The devices respond to an authentication request, e.g., from a bank accounting application, by collecting authentication samples of an ongoing conversation with the samples having sufficient detail for authentication. The devices send the authentication samples in parallel to ongoing conversation data (e.g., segmented in the signaling channel) without disrupting the conversation or violating bandwidth requirements. Authentication samples may be verified prior to authentication by comparison against the corresponding portion of the ongoing conversation. 
    
    
     
       BRIEF DESCRIPTION OF THE DRAWINGS 
       The foregoing and other objects, aspects and advantages will be better understood from the following detailed description of a preferred embodiment of the invention with reference to the drawings, in which: 
         FIG. 1  shows an example of an Internet Protocol (IP) communications system that transparently provides a voice signature of sufficient quality for voice identification and authentication during conversational communication according to a preferred embodiment of the present invention; 
         FIG. 2  shows an example of voice identification and authentication signaling, e.g., between a SIP phone and a bank application according to a preferred embodiment of the present invention; 
         FIG. 3  shows a block diagram example of an implementation of a system for carrying out authentication during a conversation with SIP device. 
     
    
    
     DESCRIPTION OF PREFERRED EMBODIMENTS 
     Turning now to the drawings and more particularly,  FIG. 1  shows an example of an Internet Protocol (IP) communications system  100 , e.g., a Voice over IP (VoIP) communications system, transparently providing voice samples and signatures of sufficient quality for voice identification and authentication during conversational communication using a typical high compression codec with corresponding low audio quality, according to a preferred embodiment of the present invention. The preferred system  100  may be a session initiation protocol (SIP) system that includes a digital call capable network  102  coupled to a state of the art voice identification and authentication system  104 , e.g., a bank, storing voice signatures, e.g., in non-volatile storage  106 . The system includes End Points (EP)  108 ,  110 ,  112  with connected digital telephony devices (e.g., VoIP phones) and Multimedia Terminal Adapters (MTA), e.g., keysets, cell phones and/or SIP phones. Since a network device defines an EP, each EP and a device(s) at the EP are referred to herein interchangeably. A gateway  114 , e.g., a state of the art media gateway, connects the network externally  115 , e.g., to a public switched telephone network/public land mobile network (PSTN/PLMN) and/or the Internet. A preferred softswitch  116  manages network EP communications. 
     Preferably, the EPs  108 ,  110 ,  112  are state of the art VoIP phones and VoIP devices, and in particular high-end VoIP devices with a high quality microphone  118 , sophisticated audio circuitry (not shown) and a local speaker  119 . Preferably also, state of the art voice identification and authentication system  104  includes one or more substantially similar state of the art VoIP phones and VoIP devices and may be directly connected to the preferred digital call capable network  102  or connected through the external network  115 , indicated by the dashed line. Also, although as described herein, each of the SIP devices  108 ,  110 ,  112  described in this example includes the requisite audio circuitry, it is understood that this audio circuitry may be included in a media gateway  114  coupling communications devices to state of the art voice identification and authentication system  104  through the external network  115  or distributed between SIP devices  108 ,  110 ,  112  and the media gateway  114 . Further, media gateway  114  provides the highest available voice data quality to state of the art voice identification and authentication system  104 . 
     While for normal VoIP communications, the EPs  108 ,  110 ,  112  use a standard telecom (e.g., Gxxx) codec to transmit live audio data, with voice quality intentionally reduced to fit into narrowband audio channels; when requested, these devices  108 ,  110 ,  112  selectively provide access to high quality voice data samples. In particular, these high quality voice data samples are of sufficient detail (e.g., sampling rate and precision) for voice used in state of the art for signatures identification and authentication, referred to herein as authentication samples. 
     For example, when the bank  104  is performing voice recognition and authentication, it requests that the respective device  108 ,  110 ,  112  transmits an authentication sample in parallel. The respective device  108 ,  110 ,  112  may avoid surpassing allocated bandwidth limits by limiting the duration of the authentication samples. Further, because they are separate from the conversation, the authentication samples need not be transmitted contemporaneously in quasi real-time, while the authentication completes in relative-time fashion, i.e., during the conversation. So, the respective device  108 ,  110 ,  112  may respond to a request by sampling audio data for a selected period of time sufficient for authentication at a selected authentication quality, and the collected sample data is spooled, e.g., in EP storage  120 , and transmitted at a relatively low rate for the volume of collected data. The authentication period and quality may be specified, for example, in the request or by default. 
     In VoIP telephony systems with signaling and media channels using separate transmission channels, authentication samples may transfer in either of these channels, or in any other available channel. Preferably, however, authentication samples transfer in the more reliable channel, e.g., signaling. Authenticity of the source of data may be ensured by requesting a random sampling of a respective conversation. Furthermore, by referencing the authentication samples against real-time audio transmissions, authenticity may be validated by the continuity of the real-time conversation itself, e.g., using typical state of the art audio content comparison methods to compare an authentication sample(s) against the corresponding real-time audio. This authenticity comparison may be initiated with a simple request signal. Further, processing such an authenticity request may be subject to mutual agreement and negotiation, e.g., by user preauthorization or by prompting for user authorization. Moreover, either or both the authentication sample(s) and the corresponding real-time audio may be encrypted using well known data encryption, in addition to or in consonance with normal network encryption. 
       FIG. 2  shows an example of voice identification and authentication signaling, e.g., between SIP phone  110  and bank  104  through digital call capable network  102  and/or external network  115  in the system  100  of  FIG. 1 , according to a preferred embodiment of the present invention. In this example, a user at SIP phone  110  calls the bank customer service (e.g., a banking application or bank accounting system  106 ) through the softswitch  116  in his/her provider network  102 , establishing a stable call talk state  122  between them. Bank customer service decides to authenticate the caller using voice authentication and so, initiates  124  sending a “hi-Quality-audio request”  126  to the softswitch  116  with a Subscribe (Hi-Quality speech, 5 sec) SIP request that requests a 5 second authentication sample in this example. The softswitch  116  forwards the SIP request  128  through network to the SIP phone  110 , while the regular ongoing audio exchange continues through a Real-time Transport Protocol (RTP) channel  130 . The SIP phone  110  responds to the SIP request  128  by beginning to collect the requested authentication sample for the next 5 seconds. Since the sample size is relatively large as compared to voice communications data, in this example, the sample is fragmented or segmented, and the segments are transferred spread over a sufficient period of time to minimize/eliminate the impact of transferring the entire sample on communications system load. 
     So, the first data segment is sent  132  to the softswitch  116  in a SIP message, a Notify (Hi-Quality:data) message. The softswitch  116  forwards the SIP message  134  to the bank  104  for bank accounting system  106 . Subsequently, remaining segments are sent in SIP messages  132 A,  132 B to the softswitch  116 , which forwards the segments  134 A,  134 B to the bank  104  for bank accounting system  106 , while the regular ongoing audio exchange continues through RTP channel  130 . It should be noted that the same RTP channel  130  is shown 3 times to indicate that the audio exchange is ongoing. Also, it should be noted that each data segment may be sent as soon as collecting it is complete with each of  132 ,  132 A,  132 B and  134 ,  134 A,  134 B being 1⅓ seconds apart for the 5 second sample on this example. Alternately, the segments may be sent at any suitable pace, and/or the entire segment may be collected, segmented and the segments sent in any order. After the requested sample has been transferred (i.e., the last segment is forwarded  134 B), the bank  104  or bank accounting system  106  may signal the termination, e.g., sending a SIP Subscribe (end of subscription) message  136  to the softswitch  116 . The softswitch  116  forwards the SIP Subscribe message  138  through network to the SIP phone  110 ; again while the regular ongoing audio exchange continues through RTP channel  130 . 
     Since the regular live audio connection is maintained through RTP channel  130  while the sample is transferred, the RTP channel  130  carries the same audio albeit at a lower quality and with different encoding. As noted hereinabove, the authentication sample and/or segments may be compared against the live audio connection to ensure that the same content is transferred over both channels to insure that, for example, a previously recorded high quality audio (e.g., an mp3) has not been substituted. 
       FIG. 3  shows a block diagram example of an implementation of the system  100  of  FIG. 1  carrying out authentication during a conversation with SIP device  112 , as in the example  FIG. 2  after having established talk state  122  and receiving the Subscribe request  128  at the SIP device  112 . As noted hereinabove, SIP device  112  is a high-end VoIP device with a high quality microphone  118 , and standard audio circuitry, an analog conditioner  140  for providing a high quality analog audio signal and a digitizer  142  for digitizing the analog audio signal. The digitized audio signal is provided both to a Gxxx codec (e.g., G729) encoder  144  for conversational coding/decoding and to an authentication encoder  146 . 
     Since authentication requires much higher quality data than conversation, the authentication encoder  146  encodes the digitized audio signal to sufficient detail (e.g., sampling rate and precision) for providing voice signatures in identification and authentication. This may be done by hardware and/or software or both. So, for example, the digitizer may provide 16 bit samples at 8K samples per second, which pass directly to authentication encoder  146  with only the most significant 8 bits being passed to G729 codec encoder  144  for every eighth sample. Alternately, the same data may be passed to both encoders  144  and  146  with the G729 codec encoder  144  applying a suitable well known compression algorithm to the digitized audio signal. 
     The authentication encoder  146  passes the encoded authentication sample (segments) to spooler  120 ; and the G729 codec encoder  144  passes conversation packets to packetizer  148 , which forwards packets to socket controller  150 . Signaling and call control  152  selectively forwards spooled segments to socket controller  150 . Socket controller  150  in the SIP device  112  establishes a stable call talk state ( 122 ) through network  102 / 115  and socket controller  154  in the bank  104  and controls regular ongoing audio exchanges through RTP channel ( 130 ) between them. The socket controllers  150 ,  154  also establish the SIP messaging channel  156 , which carries SIP requests ( 126 ,  128 ) and messages ( 132 ,  132 A,  132 B,  134 ,  134 A,  134 B,  136  and  138 ). 
     In the bank  104  the socket controller  154  forwards conversation packets to receiver  158  and signaling and call control  160  identifies authentication sample segments, which are forwarded to spooler and verification unit  162 . Receiver  158  extracts encoded conversation data from conversation packets and forwards the data to decoder  164 , which decodes the encoded conversation data. The decoded conversation data passes to both to spooler and verification unit  162  for real time comparison with sample segments and to a digital to analog (D/A) converter  166 . D/A converter  166  converts the decoded conversation data to an analog signal that is amplified by audio amplifier  168  and provided as one end of a conversation on speaker  170 . After the complete sample is verified by spooler and verification unit  162 , authentication unit  172  compares it against a stored signature from signature database  106  and provides the result  174  of the comparison as success of fail, e.g., to bank accounting system  106 . Once the authentication is complete, the authentication unit  172  signals completion ( 136 ,  138 ) through signaling channel  156 . Voice signatures may be collected substantially identical to voice authentication with the collected voice signatures stored in signature database  106 . 
     Advantageously, the present invention transparently enables voice identification, voice authentication and voice signature communications in narrowband voice communication systems, e.g., in state of the art VoIP telephony systems, while satisfying the high-compression limits of voice communications codec. 
     While the invention has been described in terms of preferred embodiments, those skilled in the art will recognize that the invention can be practiced with modification within the spirit and scope of the appended claims. It is intended that all such variations and modifications fall within the scope of the appended claims. Examples and drawings are, accordingly, to be regarded as illustrative rather than restrictive.