Patent Publication Number: US-7212502-B2

Title: Method and apparatus for dynamically adapting telephony analog loss based on channel content

Description:
FIELD OF THE INVENTION  
   This invention relates generally to a method and apparatus for implementing an adaptive analog loss plan to optimize channel performance in a communication system, and more particularly to a method and apparatus for implementing an analog loss plan in which the loss is determined in accordance with the content of the traffic being transmitted over the communication system. 
   BACKGROUND OF THE INVENTION  
   Various standards have been proposed to allow transparent bi-directional transfer of Internet Protocol (IP) traffic between the cable system headend and customer locations over an all-coaxial or hybrid-fiber/coax (HFC) cable network. Two such sets of standards have been developed, the first by the Cable Television Laboratories (CableLabs) under the PacketCable project designation, which is commonly referred to as “PacketCable.” The second standard has been developed by the European Telecommunications Standards Institute (ETSI) and is referred to as IPCablecom, TS 101 909 Digital Broadband Cable Access to the Public Telecommunications Network; IP Multimedia Time Critical Services. Among other things, these sets of standards specify a scheme for service flow for real-time services such as packet telephony. Packet telephony may be used to carry voice between telephones located at two endpoints. Alternatively, packet telephony may be used to carry voice-band data between endpoint devices such as facsimile machines or computer modems. 
   Today, access to the Internet is available to a wide audience through the public switched telephone network (PSTN). Typically, in this environment, a user accesses the Internet though a full-duplex dial-up connection through a PSTN modem, which may offer data rates as high as 56 thousand bits per second (56 kbps) over the local-loop plant. 
   However, in order to increase data rates (and therefore improve response time), other data services are either being offered to the public, or are being planned, such as data communications using full-duplex cable television (CATV) modems, which offer a significantly higher data rate over the CATV plant than the above-mentioned PSTN-based modem. Services under consideration by cable operators include packet telephony service, videoconference service, T1/frame relay equivalent service, and many others. 
   When providing any of the aforementioned services over a cable network it is necessary to reduce the level of echo signals that are present on the transmission paths. The echo signals, which are unavoidably present in each path between end users, are primarily echoes of the far end user&#39;s voice or voice-band data signals. These echo signals are carried on the transmission path to the receive end user and are reduced there based upon a so-called fixed analog loss plan. The fixed loss plan provides that a predetermined fixed amount of loss be present in the transmit and receive paths. The particular amount of loss depends on the standards to which the cable network conforms. For example, Cable Television Laboratories specifies analog loss plan of 4 dB in both the transmit and receive directions. On the other hand, the ETSI Guide specifies an analog loss plan of 4 dB in the transmit direction and 11 dB in the receive direction. 
   The need to reduce echo also arises in the context of the PSTN itself. In this case the loss plan is generally specified by the regional telephone companies and typically depends on whether the call is intra-office, intra-exchange (local), intra-LATA (toll) or inter-LATA (toll). 
   Given the wide variety of services to be deployed over various communication systems, it would be desirable to provide a loss plan that better optimizes the quality of service that can be offered by them. 
   SUMMARY OF THE INVENTION  
   The present invention provides a method for introducing analog loss to an information-bearing signal being transmitted to or from end-user terminal equipment over a communication system. The method begins by establishing a communication path to the terminal equipment over the communication system and then introducing a level of loss into the communication path to reduce echo. The level of loss that is introduced is based on the information content of the information-bearing signal. 
   In accordance with one aspect of the invention, a greater amount of loss is introduced for voice content than for voice-band data content. 
   In accordance with another aspect of the invention, the communication system is a cable network. In some cases the cable network may be a hybrid-fiber/coax cable network. In other cases the communication system may be a public-switched telephone network. 
   In accordance with yet another aspect of the invention the communication path supports packet telephony. 
   In accordance with another aspect of the invention, the communication path maybe a receive path to the terminal equipment, a transmit path from the terminal equipment, or both a receive and transmit path. 
   In accordance with another aspect of the invention, an apparatus is provided for attenuating an information-bearing signal being transmitted to or from end-user terminal equipment over a communication system. The apparatus includes a content detector for sampling the information-bearing signal to distinguish between voice content and voice-band data content and generating a control signal in response thereto. An attenuator circuit is also provided for introducing a selective level of loss into a communication path in which the information-bearing signal travels. The apparatus also includes a controller that couples the content detector to the attenuator circuit and selects the selective level of loss introduced by the attenuator circuit in response to the control signal. 
   In accordance with another aspect of the invention, a computer readable medium is provided for attenuating an information-bearing signal being transmitted to or from end-user terminal equipment over a communication system. The computer readable medium contains program instructions that, when loaded into a processor, cause the processor to perform the steps of: sampling the information-bearing signal to distinguish between voice content and voice-band data content and generating a control signal in response thereto; selecting a selective level of loss based on the control signal; and introducing the selective level of loss into a communication path in which the information-bearing signal travels. 

   
     BRIEF DESCRIPTION OF THE DRAWINGS 
       FIG. 1  shows an illustrative communications system embodying the principles of the invention. 
       FIG. 2  shows one embodiment of an attenuation network for implementing a dynamically adjustable level of loss in accordance with the present invention. 
       FIG. 3  shows an alternative embodiment of the resistor network employed in the attenuator network depicted in  FIG. 2 . 
       FIG. 4  shows an alternative embodiment of the attenuation network depicted in  FIG. 2 . 
   

   DETAILED DESCRIPTION  
   The present inventor has recognized that the analog loss plan employed in a communication system such as a cable network or the public switched telephone network (PSTN) effectively requires a tradeoff in the quality of service offered for voice traffic and voice-band data traffic. That is, loss plans are generally fixed regardless of the nature of the content of the traffic. In many cases, however, the quality of service may be enhanced by changing the level of loss based on whether the traffic is voice or voice-band data. Accordingly, as described in more detail below, in the present invention an analog loss plan is provided in which the level of loss is dynamically changed based on the content of a call. The level of loss may be adjusted in the receive path, the transmit path, or both paths. While the loss plan will be described in terms of a cable network, the dynamically adjustable loss plan is equally applicable to other communication networks such as a PSTN. 
   An illustrative communications system embodying the principles of the invention is shown in  FIG. 1 . Other than the inventive concept, the elements shown in  FIG. 1  are well known and will not be described in detail. 
   As noted above, communications system  100  is representative of a network architecture in which subscribers associated with subscriber gateways or Media Terminal Adapters (MTAs)  110   1 – 110   4  may access the Internet  175  and a Public Switched Telephone Network (PSTN)  140 . In particular, MTAs  110   1 – 110   4  are in communication with the Internet  175  via a CATV network. Cable TV network access is provided by an MSO (Multi-Service Operator) (not shown). in this context, it is assumed the MSO provides (besides the traditional CATV access network facilities exemplified by communications network  117 ) Access Node  170  and cable modem  115 . This CATV network arrangement is also referred to herein as a cable data network. CATV network is typically an all-coaxial or a hybrid-fiber/coax (HFC) cable network. MTAs  110   1 – 110   4  is also in communication with PSTN  140  via the cable network, IP network  175 , and IP Access terminal (IPAT)  130 . 
   As shown in  FIG. 1  for MTA  110   1 , the MTAs  110   1 – 110   4  include customer premises equipment  122 , e.g., terminal equipment such as a telephone or facsimile machine, a codec  128 , a Digital Signal Processor (DSP)  124 , host processor  126  and Cable Modem (CM)  115 . Terminal equipment  122 , codec  128 , DSP  124 , and host processor  126  are collectively representative of data terminal equipment, which is coupled to communications link  117  via CM  115 . CM  115  provides the access interface to the cable data network via an RF connector and a tuner/amplifier (not shown). Broadly speaking, DSP  124  generates data packets from the analog signals received from the telephone  122 . That is, DSP  124  and codec  128  collectively perform all of the voice band processing functions necessary for delivering voice and voice-band data over a cable network, including echo cancellation, packet loss concealment, call progress tone generation, DTMF/pulse and fax tone detection, audio compression and decompression algorithms such as G.723 and G.729, packet dejittering, and IP packetization/depacketization. Typically, DSP  124  encodes the data with pulse code modulated samples digitized at rates of 8, 16 or 64 kHz. Host processor  126  receives the data packet from the DSP  124  and adds an appropriate header, such as required by the MAC, IP, and UDP layers. Once the packet is complete, it is sent to CM  115 , where it remains in a queue until it is transmitted over the cable data network to the CMTS  120  in the CATV headend  170 . If the service being provided is a real-time service such as packet telephony, for example, the data packets should be formatted in accordance with a suitable protocol such as the Real-Time Transport Protocol (RTP). 
   An Internet Service Provider (ISP) provides Internet access. In the context of  FIG. 1 , it is assumed an ISP provides IP network  175 , which includes a cable data network access router (not shown) attached to communications link  132 . It should be noted that for illustrative purposes only it is assumed that the above-mentioned MSO and ISP Service provider are different entities even though this is not relevant to the inventive concept. 
   CM  115  is coupled to Access Node  170  via cable network  117 , which is, e.g., a CATV radio-frequency (RF) coax drop cable and associated facilities. Access Node  170  provides services to a plurality of downstream users (only one of which is shown) and comprises cable modem data termination system (CMTS)  120  and head-end router  125 . (CMTS  120  may be coupled to head-end router  125  via an Ethernet 100BaseX connection (not shown).) CMTS  120  terminates the CATV RF link with CM  115  and implements data link protocols in support of the service that is provided. Given the broadcast characteristics of the RF link, multiple customers and, hence, potentially many LANs may be serviced from the same CMTS interface. (Also, although not shown, it is assumed that the CATV network includes a plurality of Access Nodes.) 
   CM  115  and CMTS  120  operate as forwarding agents and also as end-systems (hosts). Their principal function is to transmit Internet Protocol (IP) packets transparently between the CATV headend and the customer location. The above noted standards have been prepared as a series of protocols to implement this functionality. 
   In a full voice-over-Internet communication system, a Call Management Server (CMS)  150  is the hardware or software component that provides the telephony intelligence in the communications system and is responsible for telephone call processing. In particular, CMS  150  is responsible for creating the connections and maintaining endpoint states required to allow subscribers to place and receive telephone calls, to use features such as call waiting, call forwarding and the like. In theory this invention can also apply to a switched IP communication system in which an IP digital terminal connected to a CLASS5 telephony switch substitutes for the CMS and IPAT. In this system, IP-based call signaling is conducted between MTA and IPDT and GR303 or V5.2 call signaling is conducted between IPDT and telephony switch. In this system IP voice traffic voice traffic is conducted between MTA and IPDT. 
   As previously mentioned, MTAs  110   1 – 110   4  include circuitry for implementing a loss plan to reduce the level of echo. Conventional loss plans do not distinguish among the nature of the content of the traffic being communicated. However, the level of loss that is optimal for voice traffic is not necessarily the same as the level of loss that is optimal for voice-band data traffic. For example, the present inventor has recognized that the amount of loss that is optimal for voice-band data traffic is generally less than the amount of loss that is optimal for voice traffic. Accordingly, it would be advantageous to vary the amount of loss that is introduced based on the content of the traffic. In the case of voice and voice-band data, for example, it will generally be advantageous to provide a lesser amount of loss for the voice-band data traffic than for the voice traffic. By continuously monitoring the content the analog loss can be optimized for the content on the transmission path in real time as the content changes. In this way there is no need to accept performance tradeoffs and the end user can be provided with the highest level of service regardless of the content being transmitted. That is, with the present invention both voice quality and modem connect speeds can be enhanced or even maximized. 
     FIG. 2  shows one embodiment of an attenuation network for implementing a dynamically adjustable level of loss in accordance with the present invention. In the communication network depicted in  FIG. 1  the attenuation will generally be provided in each of the MTAs  110   1 – 110   4 . The loss may be imparted on the digital signal at a point along its path between the cable modem  115  and codec  128  or it may be imparted on the analog signal received from (or transmitted to) the terminal equipment  122 . In the embodiment of the invention shown in  FIG. 2 , the loss is imparted on the analog signal received from a far end user after being processed by the codec  228 . In operation, the attenuation network  200  determines whether a voice-band data call or a voice call has been established. In the case of a voice call the loss is set to some pre-established level. In the case of a voice-band data call the amount of loss is reduced to better optimize transmission of the voice-band data by, for example, increasing modem connect rates, which may otherwise be reduced at high loss levels. 
   For example, in the case of the aforementioned ETSI Guide, a loss of 4dB is specified in the transmit direction and 11 dB in the receive direction. While this loss plan may be optimal for voice traffic, it is not optimal for voice-band data. Accordingly, upon detection of voice-band data, the present invention may be reduce the loss in the receive direction to a level somewhere between 11 and 4 dB. 
   Attenuation network  200  includes a content detector  202 , a control circuit  204 , and an attenuator circuit  206 . Content detector  202  samples the analog signal received from the codec  228  to determine if the signal is a data signal (e.g, a modem tone) or a voice signal. In some relatively simple implementations, content detector  202  may be an RLC or other tuned circuit that is used to detect the presence or absence of a modem tone. In other implementations the content detector  202  may detect the presence or absence of voice traffic. In general, the present invention encompasses content detectors that distinguish between voice and voice-band data by any appropriate means, including, but not limited to, either the detection of a modem tone, a voice signal or both. In the embodiment of the invention shown in  FIG. 2 , the analog signal is sampled from the 4-wire to 2-wire hybrid network  208 , which converts the 2-wire audio line from the terminal equipment into separate send and receive paths in a 4-wire audio to the codec  228 . However the content detector  202  may sample the signal at any appropriate point in the MTA between the cable modem and the terminal equipment. Accordingly, the signal that is sampled may be either a digital signal or an analog signal. 
   Once the content detector  202  has determined whether the call is a voice or voice-band data call it communicates the result to the control circuit  204 , which in turn adjusts the level of attenuation imparted by the attenuator circuit  206 . The particular attenuator circuit shown in  FIG. 2  employs a resistor network with three resistors R 1 , R 2  and R 3  and can provide two levels of attenuation depending on whether R 3  is placed in parallel with R 2 . The lower level of attenuation is determined by the ratio of R 1  to R 2  and the higher level of attenuation is determined by the ratio of R 1  to R 2 /R 3 . A transistor  210  serves as a switch that inserts and removes R 3  from the resistor network under the control of the control circuit  204 . When a call supporting voice traffic is in progress, the transistor  210  is switched so that R 3  is placed in parallel with R 2 . When a call supporting voice-band data traffic is in progress, the transistor is switched so that R 3  is removed from the circuit. Of course, the attenuator circuit can provide additional levels of attenuation by increasing the number of resistors in the resistor network. Moreover, the transistor  210  may selected from any transistor technology, including bipolar, FET or MOSFET transistors. More generally, instead of transistor  210 , any switching device may be used to adjust the overall resistance of the attenuator circuit  206 , including mechanical relays, analog transmission gates and the like. 
   The resistor network employed in the attenuator circuit  206  may be more complex than that depicted in  FIG. 2 . For example,  FIG. 3  shows a resistor ladder that can comprise any number of resistors to provide any number of levels of resistance. An appropriate arrangement of switches placed across the resistors in the resistor ladder can select the particular resistance of the resistor network that is needed to achieve the desired level of attenuation. The resistor ladder can be realized with discrete components or solid-state devices. 
   In some embodiments of the invention the attenuator network may provide a continuously variable level of attenuation instead of the discrete levels of attenuation provided by the aforementioned resistor network. For example, referring to  FIG. 4 , in this embodiment the attenuator network may employ a MOSFET transistor  408  operating in its linear mode instead of serving as a switch. A D/A converter  406  receives a signal from the control circuit  404  and generates a varying DC bias that is placed across the gate of the MOSFET  408 , which causes the drain-source resistance r ds  of the MOSFET to vary. The drain-source resistance r ds  may in effect serve as a continuously adjustable resistor that can be used to provide a continuously adjustable level of attenuation. True linear attenuation may be achieved by using a software algorithm to curve fit the MOSFET&#39;s r ds  curve and to compensate for non-linearities in the r ds  performance by modifying the drive level of the D/A converter  406  that is applied to the MOSFET  408 . 
   While the attenuation network has been described above in terms of discrete hardware components, those of ordinary skill in the art will recognize that the attenuation network may be configured in any manner known to those of ordinary skill in the art. For example, it may be implemented with either digital or analog electronics and in hardware or a combination of hardware and software. If implemented in software, the attenuation network maybe in some cases incorporated into the DSP  124  of the MTAs. 
   The present invention is also applicable to communication networks other than a cable network. For example, the invention may be employed in a PSTN. In this case the invention may be implemented in the line card that serves as the interface between the network switch and the transmission path to the end user.