3 min readfrom Machine Learning

Live Human Detector on Outbound Phone Calls [R]

Our take

Introducing the Live Human Detector for outbound phone calls—a transformative tool designed to eliminate the frustration of waiting in call center queues. By analyzing audio streams post-IVR navigation, this innovative solution detects whether your call has connected to a live agent with remarkable accuracy in under two seconds. Unlike typical automated detection tools, our system distinguishes between human speech and pre-recorded messages, ensuring a seamless user experience. Explore our related article, "Novel Problems in VLA," for deeper insights into leveraging technology for enhanced productivity.

The advent of a live human detector for outbound phone calls signifies a pivotal shift in how we manage customer interactions in call centers. The primary goal is clear: to prevent humans from idly wasting time in queue lines, a scenario that frustrates both customers and agents alike. This innovative tool aims to listen to audio streams post-Interactive Voice Response (IVR) navigation, efficiently determining whether a call has transitioned from a queue to a live representative. The implications of such technology extend beyond mere efficiency; they can profoundly shape user experience and operational productivity in the customer service sector. For a deeper understanding of how technology can streamline processes, consider exploring our articles on Efficiently filling formulas in an upper triangular table and ¿Qué negocios hacen con Excel?.

However, the challenges presented by this technology are considerable. The tool must differentiate between human speech and various other audio inputs—like pre-recorded announcements—which can often sound deceptively similar. For instance, nuances such as silence periods that accompany the transition from queue to representative can confuse the system. The complexity of audio classification in real-time necessitates high levels of confidence and precision, especially within a mere 1-2 seconds. This task is further complicated by the sophisticated nature of Text-to-Speech (TTS) engines, which make it harder to discern between machine-generated audio and authentic human interaction. The focus on machine learning to train the system using labeled data highlights the progressive approach taken in developing this technology.

The broader significance of developing a live human detector cannot be understated. It speaks to a growing trend in the integration of artificial intelligence into everyday customer service operations. As companies increasingly prioritize customer satisfaction, optimizing call handling processes could lead to faster response times and more personalized service. By empowering agents to focus on meaningful interactions rather than navigating lengthy queues, this tool not only enhances productivity but also enriches the customer experience. It’s a transformation that reflects a shift from traditional methods to innovative solutions aimed at improving both user outcomes and operational efficiency.

Looking ahead, the implications extend beyond just call centers. As this technology matures, we may witness its application in various industries, reshaping how businesses interact with customers across platforms. The ongoing exploration of frameworks and algorithms needed to enhance audio classification will be crucial in refining this technology. Questions remain about the best practices for tagging data and the potential datasets that could serve as benchmarks for training. As we engage with these developments, it will be fascinating to observe how they evolve and what new frontiers they open for customer engagement in the future. The ongoing dialogue around these innovations invites us to consider how we can further leverage AI to transform interactions, ensuring that technology remains a tool for empowerment rather than a barrier to meaningful communication.

Goal
To save humans wasting time sitting in Call Centre queues waiting to be answered

To have tool listen in on the audio stream of a live call, post IVR Navigation - to determine whether the call has transitioned out of the queue and to a live person.

Requirements

The tool must be able to classify the audio within a sub 1-2 seconds contextual window with as high confidence level as possible.

This is not a typical AMD tool, we are not just detecting machine audio vs human speech

Assumed Challenges

  1. It may be difficult to determine between a pre-recorded RVA (Recorded Voice Announcement) and a human speaking. RVA typically are professionally recorded with distinct pitches and emotional queues, have clean audio with no background noise or silence before and after the message. This is not always the case, especially if announcements are recorded in house by the general staff.
  2. When a call is transitioning and 'Answered' there is usually a distinct soft click and or some background noise before the agent starts speaking. This silence period, whilst a good indication a call has been answered could be confused with quiet periods between music or RVA announcements in the queue.
  3. It may be difficult to determine if we have been answered by Voicemail - whilst there is usually a beep at the end, the message itself would also start with a silence period followed by audio sounding similar to an RVA.
  4. A single short beep tone could mean Voicemail, Answered or it could mean the call is being recorded
  5. Identifying we are in a queue based on TTS audio may be difficult to identify as TTS engines become more sophisticated
  6. Telephony or G711a is in the frequency band of 300–3400 Hz @ 8000hz - 64 kbit/s

Approach

To train via machine leaning using labelled data, an audio classification application that analyses the acoustics, wav form or spectrograph (via Fast Fourier Transform) of the audio stream

At this stage I do not want to use STT to determine the phase or label - Although this will likely be added at a later stage as an additional layer in the pipline to increase confidence in some of these labels such as RVA/TTS/Voicemail/Call Screening

Phase

Queuing

Labels

Music, TTS, RVA (Recorded Voice Announcement)

Transitioning

Labels

Ringback, Answered, Machine Beep

Connected

Labels

Human, Fax, Voicemail, Call Screening

Disconnected

Labels

Engaged Tone

References

https://www.mdpi.com/2076-3417/12/7/3293 - YOHO You only here once
https://www.vicidial.org/VICIDIALforum/viewtopic.php?t=42330

https://huggingface.co/learn/audio-course/chapter2/audio_classification_pipeline

https://www.youtube.com/watch?v=m3XbqfIij_Y&t=32s

https://google-ai-edge.github.io/mediapipe-samples-web/#/audio/audio_classifier

https://scikit-learn.org/stable/machine_learning_map.html

https://arxiv.org/pdf/2410.08235

Question

Seeking assisance on where to actually start. Yes I be relying heavily on claude code to build this so apologies in advance

What is the best framework / algo rhythm / approach to start solving this problem. I have seen existing frameworks like YamNet work well and fast on classifying audio - however other suggest Whisper and ASR

What is the best way of tagging or labelling data. Do I label existing full length recordings with stop/start timestamps or each label or do I need to split each label into its own file - resulting in a loss of context.

Are there obvious existing data sets I should be using for some of my labels

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