VOIP voice interaction monitor

Abstract

A signal monitoring apparatus and method involving devices for monitoring signals representing communications traffic, devices for identifying at least one predetermined parameter by analyzing the context of the at least one monitoring signal, a device for recording the occurrence of the identified parameter, a device for identifying the traffic stream associated with the identified parameter, a device for analyzing the recorded data relating to the occurrence, and a device, responsive to the analysis of the recorded data, for controlling the handling of communications traffic within the apparatus.

Description

BRIEF DESCRIPTION OF THE DRAWINGS(Step 302; FIG. 3). As will be appreciated by the arrows employed for the signal lines 24, 26, the high impedance tap 20 is arranged to monitor outgoing voice signals from the call-centre 10 whereas the high impedance tap 22 is arranged to monitor incoming signals to the call-centre 10. The voice traffic on the lines 24, 26 therefore form a two-way conversation between a call-centre operative using one of the terminals 12 and a customer (not illustrated).

The monitoring apparatus 16 embodying the present invention further includes a computer telephone link 28 whereby data traffic appearing at the exchange switch 14 can be monitored as required.

The digital voice recorder 18 is connected to a network connection 30 which can be in the form of a wide area network (WAN), a local area network (LAN) or an internal bus of a central processing unit of a computer.

Also connected to the network connection 30 is a replay station 32, a configuration management application station 34, a station 36 providing speech and/or data analysis engine(s) and also storage means comprising a first storage means 38 for the relevant analysis rules and the results obtained and a second storage means 40 for storage of the data and/or speech monitor.

FIG. 2 illustrates the typical format of a data packet 42 used in accordance with the present invention and which comprises a packet header 44 of typically 48 bytes and a packet header 46 of typically of 2000 bytes.

The packet header is formatted so as to include the packet identification 48, the data format 50, a date and time stamp 52, the relevant channel number within which the data arises 54, the gain applied to the signal 56 and the data length 58.

The speech, or other data captured in accordance with the apparatus of the present invention, is found within the packet body 46 and within the format specified within the packet header 44.

The high impedance taps 20, 22 offer little or no effect on the transmission lines 24, 26 and, if not in digital form, the monitored signal is converted into digital form. For example, when the monitored signal comprises a speech signal, the signal is typically converted to a pulse code modulated (PCM) signal or is compressed as an Adaptive Differential PCM (ADPCM) signal.

Further, where signals are transmitted at a constant rate, the time of the start of the recordings is identified, for example by voltage or activity detection, i.e. so-called “vox” level detection, and the time is recorded. With asynchronous data signals, the start time of a data burst, and optionally the intervals between characters, may be recorded in addition to the data characters themselves.

The purpose of this is to allow a computer system to model the original signal to appropriate values of time, frequency and amplitude so as to allow the subsequent identification of one or more of the various parameters arising in association with the signal (see, FIG. 4). The digital information describing the original signals is then analysed at station 36, in real time or later, to determine the required set of metrics, i.e. parameters, appropriate to the particular application (Step 304; FIG. 3).

FIG. 3 is a flowchart of an example process 300 for monitoring communications traffic. At stage 302, signals representing communications traffic are monitored. For example, the digital voice recorder 18 can monitor two-way conversation traffic associated with the exchange switch 14. At stage 304, a predetermined parameter is identified by analyzing the content. For example, a digital signal processor programmed with an appropriate algorithm can identify the predetermined parameter. At stage 306, the occurrence of the identified parameter is recorded. For example, the first storage 38 (analysis rules and results) can store the occurrence of the identified parameter. At stage 308, the traffic stream associated with the parameter is identified. For example, the speech/data analysis engine 36 can identify the traffic stream. At stage 310, the recorded data relating to the occurrence is analyzed. For example, the speech/data analysis engine 36 can analyze the recorded data stored in the first storage 38.

FIG. 4 is a flowchart of an example process 400 that expands stage 304 in FIG. 3. At stage 402, a list of parameter types is determined, including: non-voice elements; the delay occurring, or the overlap between, transmissions in opposite directions; the amplitude envelope of the signals, so as to determine caller anger or episodes of shouting; the frequency spectrum of the signal in various frequency bands; ration of transmissions in each direction, the ratio of word to inter-word pauses; the language used by the speaker(s); the sex of the speaker(s); the presence or absence of particular words; the frequency and content of prosody; vibration or tremor within a voice; and the confidence/accuracy with which words are recognized to identify changes in speech patterns arising from a caller. This list may be defined in the call-centre 10 using the station 36 (speech and/or data analysis engine). At stage 404, parameters are selected from the parameter types. The selected parameters may be those that are considered appropriate to the requirements of the call-centre provider. At stage 406, an identification of one or more of the selected parameters is made. For example, the station 36 may identify parameters arising in association with the analysis of a signal being monitored.

A particular feature of the system is in recording the two directions of data transmission separately (Step 306; FIG. 3) so allowing further analysis of information sent in each direction independently (Steps 308-310; FIG. 3). In analogue telephone systems, this may be achieved by use of a four-wire (as opposed to two-wire) circuit whilst in digital systems, it is the norm to have the two directions of transmission separated onto separate wire pairs. In the data world, the source of each data packet is typically stored alongside the contents of the data packet.

A further feature of the system is in recording the level of amplification or attenuation applied to the original signal. This may vary during the monitoring of even a single interaction (e.g. through the use of Automatic Gain Control Circuitry). This allows the subsequent reconstruction and analysis of the original signal amplitude.

Another feature of the system is that monitored data may be “tagged” with additional information such as customer account numbers by an external system (e.g. the delivery of additional call information via a call logging port or computer telephony integration (CTI) port).

The importance of each of the parameters and the way in which they can be combined to highlight particularly good or bad interactions is defined by the user of the system (Step 310; FIG. 3). One or more such analysis profiles can be held in the system. These profiles determine the weighting given to each of the above parameters.

The profiles are normally used to rank a large number of monitored conversations and to identify trends, extremes, anomalies and norms. “Drill-down” techniques are used to permit the user to examine the individual call parameters that result in an aggregate or average score and, further, allow the user to select individual conversations to be replayed to confirm or reject the hypothesis presented by the automated analysis.

A particular variant that can be employed in any embodiment of the present invention uses feedback from the user's own scoring of the replayed calls to modify its own analysis algorithms. This may be achieved using neural network techniques or similar giving a system that learns from the user's own view of the quality of recordings.

A variant of the system uses its own and/or the scoring/ranking information to determine its further patterns of operation i.e.

• • determining which recorded calls to retain for future analysis,
  • determining which agents/lines to monitor and how often, and
  • determining which of the monitored signals to analyse and to what depth.

In many systems it is impractical to analyse all attributes of all calls hence a sampling algorithm may be defined to determine which calls will be analysed. Further, one or more of the parties can be identified (e.g. by calling-line identifier for the external party or by agent log-on identifiers for the internal party). This allows analysis of the call parameters over a number of calls handled by the same agent or coming from the same customer.

The system can use spare capacity on the digital signal processors (DSPs) that control the monitoring, compression or recording of the monitored signals to provide some or all of the analysis required. This allows analysis to proceed more rapidly during those periods when fewer calls are being monitored.

Spare CPU capacity on a PC at an agent's desk could be used to analyse the speech. This would comprise a secondary tap into the speech path being recorded as well as using “free” CPU cycles. Such an arrangement advantageously allows for the separation of the two parties, e.g. by tapping the headset/handset connection at the desk. This allows parameters relating to each party to be stored even if the main recording point can only see a mixed signal.

A further variant of the system is an implementation in which the systems recording and analysing the monitored signals are built into the system providing the transmission of the original signals (e.g. as an add-in card to an Automatic Call Distribution (ACD) system).

The apparatus illustrated is particularly useful for identifying the following parameters:

• • degree of interruption (i.e. overlap between agent talking and customer talking);
  • comments made during music or on-bold periods;
  • delays experienced by customers (i.e. the period from the end of their speech to an agent's response);
  • caller/agent talk ratios, i.e. which agents might be talking too much.

However, it should be appreciated that the invention could be adapted to identify parameters such as:

• • “relaxed/stressed” profile of a caller or agent (i.e. by determining changes in volume, speed and tone of speech)
  • frequency of keywords heard (separately from agents and from callers) e.g. are agents remembering to ask follow-up questions about a certain product/service etc; or how often do customers swear at each agent? Or how often do agents swear at customers?
  • frequency of repeat calls. A combination of line, ID and caller ID can be provided to eliminate different people calling from single switchboard/business number
  • languages used by callers?
  • abnormal speech patterns of agents. For example if the speech recognition applied to an agent is consistently and unusually inaccurate for, say, half an hour, the agent should be checked for: drug abuse, excessive tiredness, drunkenness, stress, rush to get away etc.

It will be appreciated that the illustrated and indeed any embodiments of the present invention can be set up as follows.

The Digital Trunk Lines (e.g. T1/E1) can be monitored trunk side and the recorded speech tagged with the direction of speech. A MediaStar Voice Recorder chassis can be provided typically with one or two E1/T1 cards plus a number of DSP cards for the more intense speech processing requirements.

Much of its work can be done overnight and in time, some could be done by the DSPs in the mediastar's own cards: It is also necessary to remove or at least recognise, periods of music, on-hold periods, IVR rather than real agents speaking etc. thus, bundling with Computer Integrated Telephony Services such as Telephony Services API (TSAPI) in many cases is appropriate.

Analysis and parameter identification as described above can then be conducted. However, as noted, if it is not possible to analyse all speech initially, analysis of a recorded signal can be conducted.

In any case the monitoring apparatus may be arranged to only search initially for a few keywords although re-play can be conducted so as to look for other keywords.

It should be appreciated that the invention is not restricted to the details of the foregoing embodiment. For example, any appropriate form of telecommunications network, or signal transmission media, can be monitored by apparatus according to this invention and the particular parameters identified can be selected, and varied, as required.

Claims

1. A signal monitoring system for monitoring and analyzing communications passing through a monitoring point, the system comprising:

a digital voice recorder (18) for monitoring two-way conversation traffic streams passing through the monitoring point, said digital voice recorder having connections (20) for being operatively attached to the monitoring point;

a digital processor (30) connected to said digital voice recorder for identifying at least one predetermined parameter by analyzing the voice communication content of at least one monitored signal taken from the traffic streams;

a recorder (38) attached to said digital processor for recording occurrences of the predetermined parameter;

a traffic stream identifier (36) for identifying the traffic stream associated with the predetermined parameter;

a data analyzer (36) connected to said digital processor for analyzing the recorded data relating to the occurrences; and

a communication traffic controller (34) operatively connected to said data analyzer and, operating responsive to the analysis of the recorded data, for controlling the handling of communications traffic within said monitoring system.

2. The monitoring system of claim 1, wherein said at least one predetermined parameter includes a frequency of keywords identified in the voice communication content of the at least one monitored signal.

3. The monitoring system of claim 1, wherein said digital processor further identifies episodes of anger or shouting by analyzing amplitude envelope.

4. The monitoring system of claim 1, wherein said at least one predetermined parameter is a prosody of the voice communication content of the at least one monitored signal.

5. The monitoring system of claim 1, wherein said connections for being operatively attached to the telephony exchange switch are attached via high impedance taps (20) to telephone signal lines (24, 26) attached to said telephony exchange switch.

6. The monitoring system of claim 1, wherein said communication traffic controller serves to identify at least one section of traffic relative to another so as to identify a source of the predetermined parameter.

7. The monitoring system of claim 1, wherein said communication traffic controller serves to influence further monitoring actions within the apparatus.

8. The monitoring system of claim 1, wherein the analyzed contents of the at least one monitored signal comprise the interaction between at least two signals representing an at least two-way conversation.

9. The monitoring system of claim 1, wherein the recorder operates in real time to provide a real-time indication of the occurrence.

10. The monitoring system of claim 1, wherein said digital voice recorder comprises an analog/digital convertor (18) for converting analog voice into a digital signal.

11. The monitoring system of claim 1, wherein said digital processor is a Digital Signal Processor (30) arranged to operate in accordance with an analyzing algorithm.

12. The monitoring system of claim 1, wherein the digital processor is arranged to operate in real time.

13. The monitoring system of claim 1, further comprising a replay station (32) connected to said digital processor and arranged such that the voice communication content of the at least one monitored signal can be recorded and monitored by said digital processor for identifying the at least one parameter at some later time.

14. The monitoring system of claim 1, wherein the at least one predetermined parameter comprises plural predetermined parameters and wherein said recorder records the occurrence of the plural predetermined parameters in each of the two directions of traffic separately.

15. The monitoring system of claim 1, wherein said traffic stream identifier comprises a means for receiving an identifier tagged onto the traffic so as to identify its source.

16. The monitoring system of claim 1, wherein said digital voice recorder for monitoring the traffic streams is operative responsive to an output from said traffic stream identifier identifying the source of the conversation in which the predetermined parameter has been identified, or a threshold occurrence of the predetermined parameter has been exceeded.

17. The monitoring system of claim 1, wherein said digital voice recorder, said digital processor, said recorder, said traffic stream identifier, and said data analyzer reside on an add-in card to a telecommunications system.

18. A method for capturing a telephone interaction, comprising:

receiving audio data packets at a switch that are transmitted over a first network, wherein the audio data packets include packet headers and packet bodies;

identifying data within the audio data packets at a data analysis engine that is communicatively connected to the switch by a second network, the identifying being based on at least one predetermined parameter associated with a payload of the audio data packets; and

recording for analysis, at a recorder, any of the received audio data packets that include the at least one predetermined parameter, wherein the recorder is communicatively connected to the data analysis engine by the second network.

19. The method of claim 18, wherein the selecting step includes identifying the traffic stream in which a particular audio data packet belongs.

20. The method of claim 18, wherein the identifying includes analyzing a packet header within the audio data packets based on a channel number, a time stamp and a data format included within the packet header.

21. The method of claim 18, wherein identifying includes analyzing a packet body within the audio data packets.

22. The method of claim 18, wherein said receiving step is active or passive.

23. The method of claim 18, further comprising analyzing, at the data analysis engine, a packet body within the audio data packets to identify voice communication content included in the audio data packets.

24. The method of claim 23, wherein identifying voice communication content includes identifying a frequency of keywords identified in the audio data packets received over the first network.

25. The method of claim 23, wherein identifying voice communication content includes identifying episodes of anger or shouting based upon an amplitude envelope associated with the audio data packets.

26. The method of claim 23, wherein identifying voice communication content includes identifying a prosody associated with the voice communication content of the audio data packets.

27. The method of claim 23, wherein the step of storing is based upon identification of voice communication content that includes a predetermined parameter.

28. The method of claim 23, wherein identifying voice communication content includes examining incoming and outgoing traffic streams to identify whether a talk-over condition exists with respect to the audio data packets.

29. The method of claim 23, wherein identifying voice communication content includes identifying whether one or more of a predetermined group of words exists with respect to the audio data packets.

30. The method of claim 23, wherein identifying voice communication content includes identifying stress voice content associated with the audio data packets.

31. The method of claim 30, wherein stress is identified by determining changes in volume, speed and tone of voice content associated with the audio data packets.

32. The method of claim 23, wherein identifying voice communication content includes identifying a delay between data packet transmissions in opposite directions.

33. A recording and analysis system, comprising:

a monitoring interface operable to receive audio data packets transmitted on a computer network, the audio data packets including a packet header and a packet body, and being associated with a two-way voice interaction;

a data analysis engine operable to analyze data from selected audio data packets, including analyzing the packet header by analyzing a channel number, a time stamp and a data format within the packet header, and analyzing the packet body; and

a storage device operable to capture at least a portion of the audio data packets responsive to the analysis module.

34. The system of claim 33, wherein the data analysis engine is configured to select audio data packets based upon predefined information.

35. The system of claim 34, wherein the data analysis engine determines which of a plurality of voice interactions to which a selected audio data packet belongs.

36. The system of claim 33, wherein the monitoring interface is an active interface or a passive interface.

37. The system of claim 33, wherein the storage device is further operable to sort the audio data packets in accordance with a timestamp.

38. The system of claim 33, wherein the data analysis engine is configured to analyze voice communication content associated with packet bodies of the audio data packets.

39. A recording system for capturing and recording audio data packets transmitted across a data network, comprising:

a data switch operable to receive a plurality of call setup requests, requesting to establish a voice data session between a calling party and a called party, the voice data session comprising audio data packets communicated between a calling party and a called party via a data network;

a monitoring device operable to capture the audio data packets received by the data switch, wherein the monitor is operable to identify a call to which the audio data packets belong, and to associate the audio data packets to a voice interaction session; and

a data store operable to interface with the monitor and to record at least a portion of the received audio data packets to a record associated with the voice interaction session.