Patent Publication Number: US-6904037-B2

Title: Asymmetric implementation of DSVD for voice/data internet access

Description:
BACKGROUND OF THE INVENTION 
   This invention relates generally to transferring voice and data over Internet systems and more particularly to an asymmetric Digital Simultaneous Voice/Data system that facilitates transmission of voice and data to different endpoints. 
   Many home users have only one phone line, and hence use Digital Simultaneous Voice/Data (DSVD) modems to simultaneously transfer voice and data at the same time over a common communication channel. DSVD modems conforming to a V.34 standard are beginning to become available at low cost and are expected to be ubiquitous in new consumer-oriented personal computers. PC hardware and software companies expect to significantly reduce support costs by providing concurrent voice and data access for help desks and support organizations. 
   Conventional DSVD operates as an “end-to-end” system. In an end-to-end system, a communication channel, such as through a Plain Old Telephone Service (POTS) telephone call, directly connects two endpoints at the physical channel level. End-to-end systems differ from “access” systems where the communication channel connects to a packet switched network such as the Internet. 
     FIG. 1  shows a typical implementation of DSVD as an end-to-end voice and data system  12 . A telephone  14  and a personal computer  16  are each coupled to a conventional communication DSVD modem  18 . The DSVD modem  18  is coupled through a Public Switched Telephone Network (PSTN) to a PBX telephone switching system  22 . A second conventional DSVD modem  24  is coupled between the PBX  22  and a telephone  26  and a personal computer  28 . 
   Referring to  FIG. 2 , each DSVD modem  18  and  24  includes a telephone interface  30  having a voice data port that connects to the telephone  14  ( FIG. 1 ) for receiving analog voice signals. A UART  34  includes a data port that connects to the personal computer  16  (FIG.  1 ). The voice codec  32  is implemented either via a DSP or software running on a microprocessor, and digitizes the analog voice signal from telephone  14 . The codec  32  then typically performs a voice compression algorithm based on a G.729 compression standard. A framer/multiplexer  36  is connected to both the codec  32  and the UART  34 . The framer/multiplexer formats the voice and data into frames and then multiplexes the voice and data frames together into a continuous data stream. The data stream is transmitted by a V.34 data modem through a Data Access Arrangement (DAA)  40  over the PSTN  20 . 
   The implementation of the DSVD modems  18  and  24  are symmetric meaning the voice data is encoded, compressed, and multiplexed with the computer character data for transmission at one end. The voice and character data is then de-multiplexed, decompressed, and decoded at the other end in a reverse manner. Such a scheme leads to a number of serious limitations on how DSVD may be employed. 
   In order to support both voice and data, symmetric DSVD requires a DSVD modem at both ends of the transmission channel. If one user has a DSVD modem and the other user only has a conventional modem used in conjunction with PC-based packet voice software, the two users cannot transmit voice data. Another substantial limitation is that DSVD systems are only capable of one physical channel switched connection at a time. Thus, in DSVD systems, the voice and data must always terminate at the same endpoint. 
   Consumers of the voice and data may not be at the same endpoint. For example, voice may be sent to a service representative for catalog company “A” while the data may be sent to a World Wide Web (WWW) site for company “B” to search for alternative pricing information. Other operations may also be performed on the transmitted voice stream, such as recording conversations, performing voice recognition, etc. However, the analog voice signal output from the DSVD modem cannot be directly processed in a digital signal processing environment. The analog voice signal would require reencoding back into a digital data format. Reencoding voice signals require additional time and signal processing circuitry. The quality of the voice signal also degrades each time the voice signal is encoded and decoded between an analog signal and digital data. 
   Accordingly, a need remains for a system that is more effective in transmitting and receiving voice and data to and from different network endpoints over the same communication channel. 
   SUMMARY OF THE INVENTION 
   Voice and data is transmitted over a public telephone line in a standard DSVD data format and then formatted into data packets. Because the voice is not decoded back into an analog voice signal, the voice and data stream output from a modified DSVD modem is interfaced directly to a network access server. The network access server is coupled to a network access system, such as Internet, for routing the voice and data packets to different endpoints. The endpoints comprise different computer and telephone systems. The telephone system receives voice packets from the access server through a separate telephony gateway. 
   The modified DSVD modem includes a packet framer that removes a conventional DSVD framing format data stream and stuffs bytes into the voice and data forming IP packets. The network access server then routes the voice and data packets to the different endpoints identified in a packet header. Since the voice and data are output from the DSVD modem as data packets, the voice and data can be routed more effectively using the data network to different endpoints. 
   When transmitting data to a conventional DSVD modem, byte stuff framing is removed from the voice and data packets. The packets are then segmented and interleaved together. DSVD framing is added to the interleaved segments forming a data stream that is transmitted via a V.34 modem over a telephone line to another DSVD modem. Because the data stream is always transmitted and received in a standard modem framing format, the system operates either with conventional DSVD modems or with the modified DSVD system described above. 
   The voice signal is no longer decoded or transcoded by the modified DSVD modem or the access server. Thus, the system overcomes limitations with conventional DSVD modems that require voice and data to terminate at the same endpoint. Voice quality is also improved since the voice signals do not require reencoding before retransmission. The improved DSVD system requires less circuitry than current DSVD modems and is, therefore, less expensive to manufacture. Cost savings multiply as the number of lines processed by an access server increases. 
   The foregoing and other objects, features and advantages of the invention will become more readily apparent from the following detailed description of a preferred embodiment of the invention which proceeds with reference to the accompanying drawings. 

   
     BRIEF DESCRIPTION OF THE DRAWINGS 
       FIG. 1  is schematic diagram of a prior art DSVD system including a conventional DSVD modem. 
       FIG. 2  is detailed block diagram of the DSVD modem shown in FIG.  1 . 
       FIG. 3  is schematic diagram of a modified DSVD system according to the invention. 
       FIG. 4  is a detailed block diagram of a modified DSVD modem according to the invention. 
       FIG. 5  is a detailed block diagram of a packet framer and interleaver for the DSVD modem shown in FIG.  4 . 
   

   DETAILED DESCRIPTION 
   Referring to  FIG. 3 , a telephone  14 , PC  16 , DSVD modem  18  and PSTN  20  are used to transmit and receive voice and data in a similar manner as that previously described in  FIG. 1. A  modified DSVD modem  42  according to one embodiment of the invention is coupled between the PSTN  20  and an access server/telephony gateway  44 . An access system  46  comprises an Intranet or Internet system that couples telephone  26  to the access server  44  through a telephony gateway  48  and couples a host computer  28  to the access server  44 . 
   The network system  46  represents any standard packet switched network system. For example, the access system  46  represents a Local Area Network (LAN) or a Wide Area Network (WAN)  44  such as Ethernet, FDDI, T1/E1, T3/E3, ISDN PRI, or ATM. The access server  44  in one embodiment is a standard packet based router system such router Model No. AS50100 manufactured by Cisco Systems, 170 West Tasman Dr., San Jose, Calif. 95134-1706. 
   Telephony gateway  48 , in one embodiment is a PC based telephony server which is commercially available from MICOM Communications Corp., 4100 Los Angeles Ave. Simi Valley, Calif. 93063. The telephony gateway includes a G.729 codec that decodes voice packets into analog voice data and dial-up circuitry for calling the telephone  26 . Alternatively, the telephony gateway  48  uses a conventional DSVD modem  18  which includes a V.34 codec for converting the voice packets into analog voice signals. The host computer  28  represents any computer system that receives and transmits network data packets. 
   At the dial-up side of the DSVD transmission, electrical voice signals are generated by telephone  14  and digital data characters are generated by PC  16 . The voice signals are converted into digital frames by the codec  32  ( FIG. 2 ) and multiplexed with the digital data characters from PC  16  by DSVD modem  18 . The DSVD modem  18  then transmits the multiplexed data stream containing the voice and data over PSTN  20  to the DSVD modem  42  which in part is the subject matter of the present invention. The DSVD modem on the dial-up end off-loads considerable computation by virtue of the G.729a voice codec  32 . 
   The multiplexed voice and data stream transmitted by DSVD modem  18  is formatted into separate voice and data packets by modified DSVD modem  42 . Of significant interest is the way in which the voice signal is maintained as digital data. Because the voice data is not converted back to an analog voice signal, both the digital character data from the PC  16  and the voice data from telephone  14  can be interfaced directly to access server  44  and then routed to different endpoints. 
   The voice packets in one embodiment are transferred to a telephony gateway  48  using an Internet protocol (IP). The IP address of telephony gateway  48  is contained in a destination address in a packet header constructed by the access server  44 . The telephony gateway converts the voice packets into analog voice signals that are sent to telephone  26 . Data packets are sent using the Internet protocol. The IP data packets include an associated IP destination address in a packet header. Access server  44  routes the data packets to the host computer  28  according to the IP destination address. 
   The destination addresses for the telephony gateway  48  and the host computer  28  are originally designated according to the phone number dialed up by telephone  14  and the IP addresses input via PC  16 . For example, a user at PC  16  enters a web page address that corresponds with host computer  28 . 
   The web page address is sent along with other character and voice data to modem  42 . The web page address is formatted into a data packet header. The access server  44  uses the address to route the packet to host computer  28 . The user dials the phone number of telephone  26  with telephone  14 . The phone number is used to establish an end-to-end connection between DSVD modem  18  and modified DSVD modem  42 . Access server  44  then converts the telephone number into an IP address associated with telephony gateway  48 . Alternatively, the PC  16  is used to select the phone number for DSVD modem  42  and destination addresses for the different voice and data endpoints. 
   While shown as two separate boxes in  FIG. 3 , the modified DSVD modem  42  and the access server/telephony gateway  44  can be combined into the same device. The access server  44  can also serve as a telephony gateway when the modem  42  is a common endpoint for voice and data. The access server/telephony gateway  44  would then convert the digital voice data into an analog voice signal. For example, dial-up participants in a computer multi-media conferencing application may require voice and data to be directed to the same host system. 
   Referring to  FIG. 4 , the system shown in  FIG. 3  uses a novel implementation of DSVD where the voice channel, rather than being decoded, is interleaved in the data stream across the interface with the access server  44  as packets. A data port  49  couples a UART  50  to the access server  44  (FIG.  3 ). A packet framer and interleaver  52  couples the UART  50  to a V.34 modem  54  and a Data Access Arrangement (DAA)  56  couples the modem  54  to the PSTN  20 . 
   The UART  50 , V.34 Modem  54 , and DAA  56  are the same components used in conventional DSVD modems. The DAA  56  provides isolation between circuitry coupled to the public phone network and is well known to those skilled in the art. The telephony interface  30  and G.729a CODEC  32  previously shown in  FIG. 2  are eliminated. The framer/multiplexer  36  shown in  FIG. 2  is replaced with a V.76 compliant packet framer and interleaver  52 . The framer and interleaver  52  separates the voice from the data and frames the data and voice as interleaved packets on the same data port channel. This channel is presented to the UART  50  so that the voice and data frames enter and leave the system as serial packets over the same data port  49 . The voice and data packets are then sent from the access server  44 , through the Internet  46  and to the endpoint  26  or  28  in a compressed packetized form. 
   Referring to  FIG. 5 , voice and data is framed and interleaved in the DSVD modem in the following manner. Data packets  62  and G.729 voice packets  64  are received byte by byte through the UART  50  from telephone  26  and host computer  28  via network  46 . The packets include byte-stuffed High-level Data Link Control (HDLC) framing conventionally used over Internet interfaces. The voice and data packets are distinguished by a Data Link Connection Identifier (DLCI) field in a packet header. A value of zero in the DLCI field identifies packets containing data and a value of one in the DLCI field identifies packets containing voice data. 
   The framer/interleaver  52  in block  66  accumulates the bytes for each voice and data packet and removes the byte-stuff framing of the network packet formatting. A segmenting block  68  breaks up each received data packet if longer than a given threshold size. The threshold data size is configurable through an interface on the DSVD modem  42  (not shown). If a packet does not require segmentation, it is sent directly to an interleaver  70 . If the packet requires segmentation, the first segment is sent to the interleaver  70  and the rest of the packet is reframed and sent to a segment buffer  72 . 
   The interleaver  70  accepts the next frame from the segment block  68  and if no segments exist in the segment buffer  72 , sends the frame to a V.76 framer  74 . If there are segmented frames in the segment buffer  72 , after sending the next frame to the framer  74 , the interleaver  70  sends one segment from the buffer  72  to the V.76 framer before accepting another voice frame from the segmenter  68 . This interleaves voice frames between the segments of data frames in a manner compatible with the segmentation scheme of the V.76 modem standard. The V.76 framer  70  performs the framing operations required to produce a V.76 compliant data stream compatible with the V.34 modem  54 . 
   Voice and data received over the PSTN  20  from a standard DSVD modem are processed by the framer and interleaver  52  in the following manner. Frames are transmitted in a multiplexed data stream  77  and received through the V.34 modem  18  in a V.76 framing format (FIG.  1 ). The V.76 multiplexed framing is removed from the data stream in block  78 . The voice and data is converted in block  76  by byte-stuffing the data into HDLC packets for transmission over network system  46 . The DLCI field in a frame header  63  is set to zero for data packets  62  and set to one for voice packets  64 . The packets are then output to the UART  50  byte-by-byte. 
   The invention is particularly attractive for servers employing internal banks of V.34 modems since little or no additional hardware is required to process the voice along with the other data. The invention also interfaces well with Voiceover-IP (VOIP) products which provide capabilities such as the telephony gateway  48  that carry the voice end-to-end over a packet switched network interfacing to standard telephone extensions, PBXs and the PSTN  20 . 
   Performance of access servers is enhanced by offloading voice coding and compression to the DSVD modem attached to the user&#39;s computer. The feature set of the access servers is expanded by providing an attractive way of accessing the network through DSVD modems. The cost of sending voice through access servers is considerably decreased since the expense of DSPs to process the voice is eliminated for those channels handling DSVD. Voice quality is also improved because the voice signal does not have to be reencoded after the voice decoding preformed by conventional DSVD modems. 
   Having described and illustrated the principles of the invention in a preferred embodiment thereof, it should be apparent that the invention can be modified in arrangement and detail without departing from such principles. I claim all modifications and variation coming within the spirit and scope of the following claims.