Abstract:
A method and apparatus to provide encryption and authentication of a mini-packet in a multiplexed real time protocol (RTP) payload. Mini-packets are assembled into a payload wherein each mini-packet includes an associated mini-header for ensuring proper processing of each mini-packet. Padding is added to mini-packets when the mini-packets are encrypted to insure each mini-packet is an integral multiple of a predetermined block size. Padding for each mini-packet is determined according to p=n−k*floor((n−1)/k), wherein p is the amount of padding added to each mini-packet, n is the actual data size, and k is the block size. The padding added to the data for each packet comprises p−1 units of padding and a final padding unit for indicating the amount of padding. An authenticator may also be added to each mini-packet. A length indicator is set in each mini-header for indicating a total length of the mini-packet including the authenticator. The authenticator may then be removed based upon knowing a type of authentication used for generating the authenticator.

Description:
BACKGROUND OF THE INVENTION 
   1. Field of the Invention. 
   This invention relates in general to Internet Protocol (IP) telephony, and more particularly to a method and apparatus to provide encryption and authentication of a mini-packet in a multiplexed real time protocol (RTP) payload. 
   2. Description of Related Art. 
   Traditionally, voice has been carried over circuit switched networks (CSN) that are designed especially for transmitting voice, e.g. PSTN and GSM. During the past twenty years, telephone systems have steadily improved and changed as businesses became dependent upon reliable communication that could overcome barriers of time and distance. As a result, enterprise-wide communications platforms have been developed to deliver a broad range of telephony services. The networking services available on these platforms include automatic least-cost routing and class-of-service routing, and applications such as voice mail, mobility and call centers. 
   During this same time period, packet switching also grew to provide reliable and easy-to-use file transfer, transaction processing and information access. Packet switching systems were first implemented as proprietary systems running over private lines. However, today packet switching has evolved into standards-based, virtual-circuit networks, e.g., frame relay and Asynchronous Transfer Mode (ATM), and the Internet. The development and wide implementation of Ethernet in the 1980s led to bridges and routers and, more recently, local area network (LAN) switching. Transfer speeds have increased, prices have decreased and there are now more than 200 million Internet and Ethernet users worldwide. 
   Currently, there is a lot of interest for the transmission of voice over packet switched networks (PSN). The next big development in telecommunications will be combining the Internet with mobile phones and other devices such as personal digital assistants (PDAs). Soon consumers will be using small communication devices that combine features such as mobile telephones, Internet terminals, music systems, video systems, cameras, etc. Further, the Internet and the growing convergence around the Internet Protocol (IP) present great opportunities for businesses to capture new markets, serve customers better, reduce costs and improve productivity. 
   The biggest challenge facing IP telephony will be accommodating business-critical applications. They include call centers, Interactive Voice Response (IVR), and other speech-activated applications, mobility and single-number roaming services, and unified messaging. 
   These types of applications accentuate the need for IP telephony to address the difficult issue of transmission quality. Over time, the telephone network has become very reliable and delivers consistently high-quality service. In contrast, on today&#39;s intranets and the public Internet, the quality of service is virtually nonexistent. File download times and the time required to pull up a web site varies, and the time for e-mail to reach its intended destination is dependent upon many network factors. Increasing the bandwidth of Internet links has been the focus of most efforts to improve the quality of service. However, increasing bandwidth is only a partial fix for the short term. In the long run, other strategies are required. 
   At present, IP networks offer a single class of service called best effort, which can not guarantee any Quality of Service (QoS) to applications. To support delay sensitive applications such as voice and interactive multimedia, there have been many proposals submitted to the Internet Engineering Task Force (IETF) on how to integrate QoS in IP networks. These proposals include differentiated service (diff-serv), Integrated services (Int-serv) and Multi Protocol Label Switching (MPLS). Despite these efforts, QoS in IP is still elusive and could take some time before it is deployed over global Internet. 
   As suggested above, IP telephony has emerged as a potential application to challenge the traditional phone companies by offering long distance telephone service over Internet for low prices. There are a large number of equipment vendors offering IP telephone gateways and accessories to provide IP telephony service to corporate customers and Internet Service Providers (ISPs). IP telephone standards such as H.323, H.225 and H.245 have been standardized to enhance the rapid deployment of IP telephone services in the global Internet. Even though, IP telephone is not a reality in the public Internet today, it has been more successful in Intranet and Virtual Private Networks (VPN) environments. 
   In trials, IP telephone services have been demonstrated to have the potential to match the voice quality offered by traditional telephone networks. As a result, the growth of IP telephone gateways in corporate and ISP environments is expected to increase exponentially in the coming years. IP telephone gateways act as an interface between the existing PSTN and PBX networks and IP networks. This method allows one PSTN user to call another PSTN user connected through IP telephone gateways. thus eliminating the need for long distance telephone network. 
   In a IP telephony connection, two sides of the PSTN/PBX users (two branches of the same company) are interconnected by IP telephone gateways. In such application, a telephone call between PSTN/PBX users located at either side of the gateways is carried by a separate Real-time Transport Protocol/User Datagram Protocol/Internet Protocol (RTP/UDP/IP) connection. RTP is an Internet protocol for transmitting real-time data such as audio and video. RTP itself does not guarantee real-time delivery of data, but it does provide mechanisms for the sending and receiving applications to support streaming data. Typically, RTP runs on top of the UDP protocol, although the specification is general enough to support other transport protocols. The User Datagram Protocol is a connectionless protocol that, like TCP, runs on top of IP networks. Unlike TCP/IP, UDP/IP provides very few error recovery services, offering instead a direct way to send and receive datagrams over an IP network. 
   IP telephony gateways provide an interface between the existing circuit switched telephone networks (such as PSTN and GSM) and the packet switched IP data networks. In traditional IP telephony applications, telephone calls between PSTN users interconnected by a pair of IP telephony gateways to compress incoming PSTN speech samples generate packets with sizes ranging from 5 to 20 bytes per speech sample. 
   For example, G.723.1 (the most popular IP telephony codec and the International Multimedia Teleconferencing Consortium&#39;s (IMTC) Voice over IP (VoIp) mandatory low bit-rate codec), generates a 20 byte speech packet at 30 ms intervals. Many codecs used in cellular environment generate less than 10 byte packet per speech sample. Small size packets are subjected to large overhead when transferred using the Real time Transport Protocol (RTP). The RTP/UDP/IP overhead is 40 bytes (12+8+20) for a simple speech packet. For example, a 10 byte packet transferred via RTP/UDP/IP increases the overhead to 80% (40 byte overhead/50 byte overhead plus packet). In addition, for each call request a single UDP/IP connection (a pair of UDP ports) is established between the gateways requiring a large state (memory) to be maintained at the telephony gateways, thereby making these less scaleable. 
   Congestion in IP networks results in packet loss at routers and UDP does not have any retransmission mechanism to recover lost packets. Also, real time applications such as speech is intolerant to delay caused by retransmission. In traditional RTP method, each individual speech packet is transmitted as a IP packet, which generates a large number of packets between gateways. This heavy traffic volume is a potential situation for congestion and packet loss at IP routers. 
   To overcome these this problem, an efficient real-time transport protocol multiplexing method and apparatus for transporting compressed speech between IP telephony gateways has been proposed in co-pending and commonly-assigned U.S. patent applications Ser. No. 09/137,276, by Baranitharan Subbiah, entitled “METHOD AND APPARATUS FOR PROVIDING EFFICIENT USER MULTIPLEXING IN A REAL-TIME PROTOCOL PAYLOAD FOR TRANSPORTING COMPRESSED SPEECH BETWEEN IP TELEPHONY GATEWAYS, herein referred to as “Subbiah.” Subbiah describes a protocol that eliminates bandwidth usage inefficiencies in transporting short packets between nodes connected by an IP network, wherein the method and apparatus enables a number of users to share a single RTP/UDP/IP connection. The protocol includes creating a header for a plurality of data packets, each header providing identification of a user associated with a packet, adding each header to the data packet associated therewith to form mini-IP payloads, multiplexing the mini-IP payloads into a RTP payload and transmitting the RTP payload over a single RTP/UDP/IP connection. 
   While Subbiah disclose a method and apparatus for providing application layer multiplexing in IP networks to overcome the problem of high header overhead for packets with small payloads, some problems remain. For example, within a multiplexed RTP packet, several mini-IP packets are multiplexed into the same RTP payload. Each of these mini-IP packets may belong to the same stream, or, more likely, to different steams. However, it is desirable to provide different security services to each of the mini-IP packets. 
   Block encryption schemes require that the packet size be an integral multiple of block size. If a packet size does not equal an integral multiple of the block size then padding bytes need to be added to the packet. The block size itself is dependent on the encryption algorithm being used. For instance, the block size in DES is 64 bits. Block based symmetric encryption encompasses the most common schemes in use today. These include DES, triple-DES, IDEA, Blowfish etc. When such schemes are used in certain modes (ECB, CBC), they require the input to be an integral multiple of the block size. 
   Currently, there exist mechanisms for providing encryption at the IP level and at the RTP level. These mechanisms have taken into account the fact that block encryption schemes require the input to be an integral multiple of the block size. This has been made possible by suitable padding schemes. However, in an environment where several mini-packets are multiplexed into an TRP packet, no suitable encryption (and corresponding padding) mechanism has been proposed. Since the different mini-packets may belong to different streams, it may be desirable to apply different encryption schemes to the different mini-packets. Similarly, there exist mechanisms for authentication of packets at the IP level, but no such mechanism has been proposed to date for authentication at the mini-packet level. 
   It can be seen then that there is a need to provide padding and encryption on a mini-packet basis. 
   It can also be seen that there is a need for a mechanism to perform padding and encryption at the mini-packet level. 
   It can also be seen then that there is a need for a mechanism to perform authentication at the mini-packet level. 
   SUMMARY OF THE INVENTION 
   To overcome the limitations in the prior art described above, and to overcome other limitations that will become apparent upon reading and understanding the present specification, the present invention discloses a method and apparatus to provide encryption and authentication of a mini-packet in a multiplexed real time protocol (RTP) payload. 
   The present invention solves the above-described problems by providing a mechanism to perform padding, encryption and authentication at the mini-packet level. 
   A system in accordance with the principles of the present invention includes assembling mini-packets into a payload wherein each mini-packet includes an associated mini-header for ensuring proper processing of each mini-packet and adding padding to mini-packets when the mini-packets are encrypted to insure each mini-packet is an integral multiple of a predetermined block size. 
   Other embodiments of a system in accordance with the principles of the invention may include alternative or optional additional aspects. One such aspect of the present invention is that padding for each mini-packet is determined according to p=n−k*floor((n−1)/k), wherein p is the amount of padding added to each mini-packet, n is the actual data size, and k is the block size. 
   Another aspect of the present invention is that the padding added to the data for each packet comprises p−1 units of padding and a final padding unit for indicating the amount of padding. 
   Another aspect of the present invention is that the unit is bytes. 
   Another aspect of the present invention is that the present invention further includes adding an authenticator to each mini-packet. 
   Another aspect of the present invention is that the present invention further includes setting a length indicator in each mini-header for indicating a total length of the mini-packet including the authenticator. 
   Another aspect of the present invention is that the invention further includes removing the authenticator based upon knowing a type of authentication used for generating the authenticator. 
   Another aspect of the present invention is that the type of authentication comprises HMAC-SHA1 and the authenticator is 20 bytes. 
   Another aspect of the present invention is that the type of authentication comprises HMAC-MD5 and be authenticator is 16 bytes. 
   These and various other advantages and features of novelty which characterize the invention are pointed out with particularity in the claims annexed hereto and form a part hereof. However, for a better understanding of the invention, its advantages, and the objects obtained by its use, reference should be made to the drawings which form a further part hereof, and to accompanying descriptive matter, in which there are illustrated and described specific examples of an apparatus in accordance with the invention. 

   
     BRIEF DESCRIPTION OF THE DRAWINGS 
     Referring now to the drawings in which like reference numbers represent corresponding parts throughout: 
       FIG. 1  shows an application scenario in which two sides of the PSTN/PBX are interconnected by IP telephone gateways; 
       FIGS. 2   a-b  illustrates mini-header according to the present invention; 
       FIG. 3  illustrates the assembly of mini-packets into a single RTP/UDP/IP payload; 
       FIG. 4  illustrates a flow chart illustrating the mini-packet protocol for providing encryption, padding and authentication to mini-packets according to the present invention; 
       FIG. 5  illustrates a padded mini-packet according to the present invention; 
       FIG. 6  illustrates a mini-packet after authentication has been performed; 
       FIG. 7  illustrates the location of the mini-packet controller in a IP layered model; and 
       FIG. 8  illustrates a mini-packet controller according to the present invention. 
   

   DETAILED DESCRIPTION OF THE INVENTION 
   In the following description of the exemplary embodiment, reference is made to the accompanying drawings which form a part hereof, and in which is shown by way of illustration the specific embodiment in which the invention may be practiced. It is to be understood that other embodiments may be utilized as structural changes may be made without departing from the scope of the present invention. 
   The present invention provides a method and apparatus to provide encryption and authentication of a mini-packet in a multiplexed real time protocol (RTP) payload. The present invention provides a mechanism to perform padding and encryption at the mini-packet level. 
     FIG. 1  shows an application scenario  100  in which two sides of the PSTN/PBX  100 ,  112  (two branches of the same company) are interconnected by IP telephone gateways  120 , 122 . In such an application, a telephone call between PSTN/PBX users  110 ,  112  located at either side of the gateways  120 ,  122  is carried by a separate RTP/UDP/IP connection. The codecs used at the telephone gateway to compress incoming PSTN/PBX voice calls generates packets with a size ranging from 5 to 20 bytes. 
   For example, the IP telephone standard G.723.1 specifies a codec that generates a 20 byte packet at the interval of 30 ms speech sample. Many codecs used in cellular environments generate a small packet, e.g., on the average a 10 byte packet per speech sample. This small size packets require a large overhead when they are transferred using the Real time Transport Protocol (RTP). 
   The RTP/UDP/IP overhead is 40 bytes (12+8+20) for each speech packet. For example, if a 10 byte packet is transferred via RTP/UDP/IP then the overhead is 80%, i.e., 40/50. In addition, for each call request a single UDP/IP connection is established between the gateways  120 ,  122  requiring a large number of states (memory) to be maintained at the telephone gateways  120 ,  122 . 
   Congestion in IP networks results in packet loss at routers and UDP does not have any retransmission mechanism to recover lost packets. Also, real time applications such as speech are intolerant to delay caused by re-transmission. In a traditional RTP method, each individual speech packet is transmitted as a IP packet, which generates a large number of packets between gateways. This heavy traffic volume is a potential situation for congestion and packet loss at IP routers. 
   The large overhead to transfer a small packets (compressed speech) through RTP/UDP/IP has been one of the drawbacks of IP telephone. In order to minimize the overhead, RTP/UDP/IP header compression is applied for slow speed links. However, this method requires compressing/ decompressing at routers as well as some additional processing overhead. 
     FIGS. 2   a-b  and  3  illustrate the use of mini-headers to reduce header overhead according to the mini-packet protocol. Even without compression, an equal or better bandwidth efficiency may still be achieved. Overhead is reduced by multiplexing two or more (e.g., up to 256) low bit rate connections in a single RTP/IP/UDP connection using a mini-header  202  as illustrated in  FIG. 2   a . Alternatively, overhead may be reduce using the mini-header  204  illustrated in  FIG. 2   b . However, those skilled in the art will recognize that the present invention is not meant to be limited to the particular mini-headers illustrated in  FIGS. 2   a-b , but that the mini-headers  202 ,  204  illustrated in  FIGS. 2   a-b  are presented for illustration only. Rather, those skilled in the art will recognize that the mini-headers  202 ,  204  enables multiplexing of multiple small size packets, and is added to each mini-packet before it is assembled with other mini-packets as an RTP payload, as illustrated in FIG.  3 . 
   To identify a single user among the number of users sharing the RTP connection, each user is allocated an unique Channel Identifier (CID) which is negotiated during connection setup. The CID negotiation procedures may be carried out by mini-packet signaling, which uses a TCP/IP connection for reliable transport. The most suitable application scenarios for mini-packet method include IP telephone gateways connecting PSTN/PBX/GSM users. 
   To identify mini-packets multiplexed on a single RTP payload, the mini-packet protocol uses a two byte header, called mini-header, for each mini-packet. The mini-header  202 , as shown in  FIG. 2   a  includes a Channel Identifier (CID)  210 , a Length Indicator (LI)  212 , and a Sequence Number (SN)  214 . The mini-header  202  allows many users to share a single RTP/UDP/IP connection thus reducing the RTP/UDP/IP overhead per packet. 
   As illustrated in  FIG. 2   a , the mini-header includes a CID field  210 , which identifies a single user among users haring a single RTP/UDP/IP connection. A CID  210  is assigned at the time of the request for access to the IP network and it is unchanged throughout the connection time. The length of the CID field  210  is 8 bits, which limits the number of users per single RTP connection to 256. 
   The LI field  212  indicates the size of the payload (speech packet) and the 6 bits allow a maximum of 64 byte payload. The variable size of the LI field  212  allows different codecs to share a single connection and offers the flexibility to transport any low bit rate connection using the mini-packet method. The size of the LI field  212  is limited to 64 bytes since most of the codes available today (G.723.1, G.729) generates packets less than 20 bytes per speech sample. 
   The 2 bit Sequence Number (SN) field  214  is used for marking the voice packets transmitted from a single user in modulo 4 method, which can be used at the receiver to identify any packet loss. The module 4 scheme will be able to identify up to 3 consecutive packet losses at IP layer. 
   The mini-header  204 , as shown in  FIG. 2   b  includes a Channel Identifier (CID)  210 , a Length Indicator (LI)  212 , a Transition bit (T)  216  and a Reserved bit (X)  218 . The Channel Identification (CID)  210  in  FIG. 2   b  is an 8 bit field which allows a maximum of 256 users to share a single RTP/UDP/IP connection. When the total number of users exceeds 256, a new RTP/UDP/IP connection is established. The LI field  212  is a 6 bit field which allows a maximum payload size of 64 bytes. The Transition bit (T)  216  is used to identify any change in processing that was applied to a mini-packet. Notification of such changes occurs by toggling the bit. Finally, the Reserved bit (X)  218  is currently undefined, but may be used, for example, as an indication of a header extension and Dual Tone Multi-Frequency (DTMF). 
   As mentioned above, those skilled in the art will recognize that the above illustration of mini-headers is not meant to limit the invention, but that other mini-header configurations and sizes could be used in accordance with the present invention. For example, the length of the fields could be modified within the 2 byte format. Further, other fields could be substituted and the length of the fields is not meant to be limited to provide a mini-header of 2 bytes. For example, the reserved bit illustrated in  FIG. 2   b  may be set to “1” to indicate an extension head is included in the mini-header thereby providing an overall length for the mini-header of 3 bytes. Alternatively, the reserved bit may be set to “0” to indicate that an extension header is not included in the mini-header. Nevertheless, those skilled in the art will recognize that increases in the overall size of the mini-header will proportionally increase the total overhead when multiple mini-packets are multiplexed together in accordance with the invention. Thus, those skilled in the art will recognize that any mini-header that enables multiplexing of multiple small size packets, is added to each mini-packet before it is assembled with other mini-packets as an RTP payload as illustrated in  FIG. 3 , and which allows proper processing of the multiple mini-packets may be used without departing from the scope of the present invention. 
   The assembly of mini-packets into a single RTP/UDP/IP payload  300  is shown in FIG.  3 . The mini-packets  330 ,  350 ,  370  follow the IP header  310 , the UDP header  312  and the RTP header  314 . Each mini-packet  330 ,  350 ,  370  is delineated by two byte mini-headers  320 ,  340 ,  360 , respectively. This approach requires a simple de-multiplexing algorithm at a receiver. Because the mini-headers  320 ,  340 ,  360  in the RTP payload  300  are transparent to the intermediate IP routers, IP packet forwarding and other functionality at the IP layer may be performed without any problems. As noted in  FIG. 3 , each of the Mini-packets  330 ,  350 ,  370  is encrypted with a distinct encryption program. The need for providing authentication and encryption at the mini-packet level arises from the need to provide end-to-end authentication and encryption. However, since mini-packets can be switched at intermediate points in the network, providing authentication and encryption at the IP level or at the RTP packet level will not provide end-to-end authentication or encryption. 
     FIG. 4  illustrates a flow chart  400  illustrating the mini-packet protocol for providing encryption, padding and authentication to mini-packets according to the present invention. First, a decision is made as to whether the mini-packet is encrypted  410 . If the mini-packet is encrypted  420 , padding is added. If the input (actual data) is of size “n” and the block size is “k”, then the amount of padding “p” is given by:
   p=n−k* floor(( n− 1)/ k ) 
   It is seen that the number of padding bytes “p” varies from one to k.  FIG. 5  illustrates a padded mini-packet  500  according to the present invention. In  FIG. 5 , the mini-packet  510  includes a data block  512 . Padding of p−1  522  is added. Even for the case where the mini-packet size equals an integral multiple of the block size, k, padding equaling one block is added. In any case, the last padding byte  524  indicates the number of padding bytes. The p−1 padding bytes  522  could be arbitrarily chosen. The endpoints of the security association are aware of the encryption mechanism and parameters. The recipient after decrypting the mini-packet looks at the last byte  524  to determine the number of padding bytes  522  used. 
   Referring again to  FIG. 4 , if encryption is not implemented  440 , or after padding is added  430 , an authenticator is added to each mini-packet for authentication.  FIG. 6  depicts a mini-packet  600  after authentication has been performed. In  FIG. 6 , the mini-packet includes a data block  610 . Authentication of a mini-packet can be done by suitably appending the authenticator  620  to each mini-packet. The length indicator in the header (not shown) of each mini-packet would indicate the total length of the mini-packet including the authenticator  620 . The recipient, upon receiving the mini-packet can separate the authenticator  620  from the actual data  610  based upon knowledge of the algorithm used for generating the authenticator. For instance, if HMAC-SHA1 was used, the authenticator  620  is 20 bytes, and if HMAC-MD5 is used, the authenticator  620  is 16 bytes. 
   A mini-packet controller is included in a gateways to implement this mini-packet method. The location of the mini-packet controller  710  in a IP layered model  700  is. shown in FIG.  7 . The mini-packet controller  710  is inserted between the IWF function (not shown) of the PSTN/GSM/PBX network  720  and the RTP module  722  in the IP telephone gateway. The mini-packet controller  710  is capable of receiving connection request from PSTN/GSM/PBX users  720  and setting up a channel on an existing or a new RTP/UDP/IP connection. The mini-packet controller  710  acts as an application to the layers below  722 ,  724 ,  726 ,  728  (RTP/UDP/IP) and uses the bearer services offered by the lower layers for effective multiplexing. Other functions of the controller include open and close RTP/UDP connections, keep track of the active users on all UDP connections, and provide inter-working with PSTN/GSM/PBX call control features. 
     FIG. 8  illustrates a mini-packet controller  800  according to the present invention. In  FIG. 8 , a demultiplexer/disassembler  810  receives RTP payloads and demultiplexes the mini-packets for analysis and control by the control and signaling module  820 . mini-packets which destined for transmission via the same output port are multiplexed again into a RTP payload at the assembler  830 . 
   Accordingly,  FIG. 8  illustrates one example of a hardware environment for the method according to the present invention. The present invention is typically implemented in the controller and signaling module  820 . The controller and signaling module  820  includes microprocessor  822  and memory  824  and other standard components. The controller and signaling module  820  executes one or more computer programs  826  which may be stored in memory  824 . The present invention comprises a method and apparatus to provide encryption and authentication of a mini-packet in a multiplexed real time protocol (RTP) payload that is preferably implemented in the controller and signaling module  820  via computer programs  826 . 
   Generally, the computer programs  826  executed by the controller and signaling module  820  may be tangibly embodied in a computer-readable medium or carrier, e.g. one or more of the fixed and/or removable data storage devices  814 , or other data storage or data communications devices. The computer programs  826  may be loaded from the data storage devices  814  into the memory  824  for execution by the microprocessor as discussed above. The computer programs  826  comprise instructions which, when read and executed by the microprocessor  822 , causes the controller and signaling module  820  to perform the steps necessary to execute the steps or elements of the present invention. Although an exemplary computer system configuration is illustrated in  FIG. 8 , those skilled in the art will recognize that any number of different configurations performing similar functions may be used in accordance with the present invention. 
   The method referred to in this description are typically stored as digital information on a computer readable and writeable medium. Many different storage mediums, such as magnetic tape and discs may be used with the invention. The data structure consists of one or more bytes of information stored on the medium. Typically, the method may consist of from several thousand to many million or billion bytes, depending on the application in which the invention is used. 
   In summary, the present invention provides a mechanism to provide encryption and authentication/integrity to a multiplexed RTP packet on a stream by stream basis. Such service differentiation among streams is critical for providers who may wish to provide value-added by differentiating security. 
   The foregoing description of the exemplary embodiment of the invention has been presented for the purposes of illustration and description. It is not intended to be exhaustive or to limit the invention to the precise form disclosed. Many modifications and variations are possible in light of the above teaching. It is intended that the scope of the invention be limited not with this detailed description, but rather by the claims appended hereto.