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AIMS TALK: Intelligent Call Center Support in Bangla Language with Speaker
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 2021 Emerging Technology in Computing, Communication and Electronics (ETCCE)
AIMS TALK: Intelligent Call Center Support in
Bangla Language with Speaker Authentication
2021 Emerging Technology in Computing, Communication and Electronics (ETCCE) | 978-1-6654-8364-3/21/$31.00 ©2021 IEEE | DOI: 10.1109/ETCCE54784.2021.9689831
Shehan Irteza Pranto Rahad Arman Nabid Ahnaf Mozib Samin
AIMS Lab, UIU AIMS Lab, UIU
Dhaka, Bangladesh Dhaka, Bangladesh
shehanirteza@gmail.com ran.nabid@gmail.com asamin9796@gmail.com
Farhana Sarker Mohammad Nurul Huda
Dept. of CSE Dept. of CSE
University of Liberal Arts Bangladesh United International University
Dhaka, Bangladesh Dhaka, Bangladesh
farhana.sarker@ulab.edu.bd mnh@cse.uiu.ac.bd
Abstract—Call support centers operate over the telephone,
connect between customers and receptionists to ensure cus-
tomer satisfaction by solving their problems. Due to pandemics,
customer call support centers have become a popular way of
communication that has been used in different domains such as
e-commerce, hospitals, banks, credit card support, government
offices. Moreover, humans’ limitations to serve 24 hours a day
and the fluctuation of waiting time makes it more challenging to
satisfy all the customers over call center. So, customer service
needs to be automated to handle customers by providing a
domain-based response in the native language, especially in a
developing country like Bangladesh, where call support centers
are increasing. Although most people use the Bangla language
to communicate, little work has been done in customer care
automation in the native language. Our developed architecture,
“AIMS TALK” can respond to that customer’s need by rec-
ognizing users’ voices, specifying customers’ problems in the
standardized Bangla language, collecting customers’ responses
to the database to give feedback according to the queries.
Besides, the system uses MFCC feature extraction for speaker
recognition with an average accuracy of 94.38% on 42 people
in real-time testing, an RNN-based model for Bangla Automatic
Speech Recognition (ASR) with a word error rate (WER) of
42.15%, and sentence summarization we used Sentence similarity
measurement technique having an average loss of 0.004. Lastly,
we used gTTS that works as Text to Speech Synthesis for the
Bangla language in WavNet architecture.
Index Terms—Text to Speech Synthesis(TTS), Automatic
Speech Recognition, Mel-Frequency Cepstral Coefficients
I. I NTRODUCTION
Phone call is the preferred top customer support channel for
almost all e-commerce companies. Customers try to resolve
their issues and queries via phone than any other medium
This research work is funded by ICT innovation Fund (a2i), ICT Division,
Ministry of Posts, Telecommunications and Information Technology, the
People’s Republic of Bangladesh
978-1-6654-8364-3/21/$31.00 ©2021 IEEE
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Nabeel Mohammed
AIMS Lab, UIU Dept. of CSE
Dhaka, Bangladesh North South University
Dhaka, Bangladesh
nabeel.mohammed@northsouth.edu
Khondaker A. Mamun*
Dept. of CSE
United International University
Dhaka, Bangladesh
mamun@cse.uiu.ac.bd
such as email, social media, or chat. Customer satisfaction
over telephone calls depends on the behavior of customer
service employees. Moreover, the behavior of customer service
employees depends on their mood; sometimes, biases and
unresponsiveness may occur. In fact, human service members
cannot support 24 hours customer services. So customers can-
not get support from the hot-line after office time. Moreover,
due to pandemics, the issues of traditional customer call center
service are becoming apparent as an increasing number of
people are trying to get service through call centers.
With the increasing difficulty of serving customers, now
artificial intelligence (AI) can be the scalable and most cost-
efficient solution for improving customer service [2]. Accord-
ing to the latest report of smallbiztrends titled, “Local Business
Websites and Google My Business Comparison Report” said,
60% of the customers prefer to call over the phone during
pandemic instead of visiting physical shop [14]. Issues include
limited service available only in the office period, sluggish
service that led to the increase of customers’ waiting time,
and service delay during peak period [4]. Studies show that
66% of the customers prefer to solve their issues within 10
minutes, or they may readily switch to alternative service [1].
AI can automate resolution while cutting costs and enhancing
customer satisfaction, allowing human agents to work on
more complex issues [3]. Many E-commerce companies have
started to implement various forms of AI to understand their
customers better and provide an improved customer experience
but working with the Bangla language has limited progress. In
addition, around 37% of e-commerce customers use automated
management services such as chatbots to react quickly in an
emergency situation to support 24/7 customer [1].
Researchers have been working on finding the best possible
solution integrating AI-assistant with human customercare
employees to ensure user satisfaction through automated call
center services. Feng-Lin Li et al. [18] presented AliMe Assist,
which is mainly designed for providing automatic online shop-
ping services in the E-commerce domain that addresses 85%
of questions per day from customers. In another study, Linjian
Sun et al. [20] proposed a service robot, where various types
of deep learning-based independent modules such as image
classification and human-computer interaction are integrated to
automate customer service. In a different study, Hilda Hardy
et al. [21] presented a dialogue management system, which
can interact with a user in spoken language having Auto-
matic Speech Recognition, Text-to-Speech conversion, dia-
logue manager, and information database to provide telephone-
based automatic customer service. Furthermore, G. Zweig et
al. [22] proposed an automated quality monitoring system for
call centers that combined the Speaker Recognition module
with maximum entropy classification, pattern recognition tech-
nology, and automatic speech recognition to develop the robust
set-up. Afterward, Graeme Mclean et al. [23] conducted a web-
based survey with 302 respondents on real-life chat service
experiences to understand the influencing parameters involved
in customer satisfaction. A.K. Warnapura et al. [24] proposed
an artificial intelligence-based architecture, where a particular
customer can get various types of information in the form of Figure 1: System Architecture of Intelligent Call Center inte-
texts/voices or emails. They used sentiment analysis for classi- grating with Speaker and Interaction Recognition
fying the response of the users, Natural Language Processing
(NLP), and Automatic Speech Recognition to automate the
whole robust system. Furthermore, Aderemi A. Atayero et II. M ETHODOLOGY
al. [25] implemented an automated speech-enabled customer
care service system called “ASR4CRM”. They developed the This section illustrates the integration of various types
system three tire architecture: telephone system, gateway and of deep learning and Natural Language Processing (NLP)-
deployment in web server, and database management system. based technologies to develop the robust architecture of AIMS
Our proposed AIMS TALK connects the bridge between TALK. At first, after receiving a user’s call, the system will
the latest architecture customized for Bangla native speakers, ask the users’ name and date of birth using Text-to-Speech
though this architecture can be used in other languages. The (TTS) synthesis. Afterward, the system extracts the MFCC
architecture of AIMS TALK uses MFCC feature extraction features from the user’s voice and sends them to the GMM
and a GMM-based adaptation model for speaker recognition to based speaker recognition module. If the speaker recognition
differentiate the authorized and unauthorized users, having an module recognizes the voice, the system will give a welcome
accuracy of 96.27% for 42 samples in real-time testing. Total message to the user in a spoken form and ask the reason of
architecture has three different databases: Personal Informa- calling in the call center. Next, the system will receive the
tion database, Generic Information database, and Credential response from the user, and using ASR, it will convert into the
Information Database. As the model can deliver credential text and save it into the personal information database where
information to the users, we used a two-step authentication the personal details have been stored. Afterward, the system
process to provide the best security. Besides, the new users will apply the Sentence Summarizing technique and extract
can only get access to the response of the FAQs from the the valuable information from the user response, encode it
generic information database. Moreover, for Automatic Speech with a sentence transformer, and identify the question type
Recognition in Bangla language, we use “Deep Speech 2” from the five different types of groups of questions prepared
[19] model gives 42.15% of Word Error Rate (WRE) on in the credential information database. Then the system will
annotated Bangla speech corpus- ’Large Bengali-Automatic ask the user some predefined questions based on that particular
Speech Recognition Training Data’ (LB-ASRTD) by google. group of questions. After identifying the groups of question
The captured response is summarized by the seq2seq model sets from the user response, the system will receive the specific
in Bangla language [16] having average testing loss is 0.004, questions from the user and give the response in spoken
which is used for specifying the domains of problems by form according to our developed credential database, where
similarity check from our developed databases. Our system similarity checking will be used to identify the particular
uses the WavNet algorithm by gTTS in the Bangla language question from our database.
to communicate with users for real-time response. On the other hand, if the speaker recognition module doesn’t

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 Figure 2: Workflow of Speaker Recognition module
recognize the voice of the user, the system will ask some extra
security questions (National Identity No, Mobile no.) to check
into the personal information database (PID) whether the user
is previously registered in the system or not. If the system finds
the user information in the PID, the previous method will be
applicable. In contrast, the system will adapt the voice features
of the user with a GMM-based algorithm and save it with
their information into the PID. Afterward, similarly, the system
will ask the reason for calling, convert it into text form using
ASR, save it into the PID, apply Sentence Summarization, and
encode it with sentence transformer. Then after the similarity
checking with the generic information database, the system
will respond with the most appropriate answer to the user from
the predefined knowledge-based generic information database.
At last, the system will ask the user to rate their satisfaction
level after using the system, and it will refer a human assistant
if it finds the rating low.
A. Speaker Recognition
We used the MFCC feature extraction [5] technique to
recognize the speaker from the audio signal. These MFCC
features are low-frequency components. Besides, the feature
matching technique integrated with the maximization algo-
rithm is created by using Gaussian Mixture Model (GMM) [7].
For implementing the speaker adaptation, our system collects
20 seconds of voice data from the user and uses feature
extraction techniques to create a GMM model of each speaker
to perform similarity matching. Furthermore, the threshold
mechanism identifies the speakers who are not registered in
the database, significantly improving the system’s efficiency by
avoiding the wrong prediction. Fig. 2 illustrates the workflow
of the Speaker Recognition module.
B. Automatic Speech Recognition
We use an ASR model to convert consumer speech into
text. Using a publicly available annotated Bangla voice corpus
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Figure 3: Sentence Summarizing Architecture
- “Large Bengali-Automatic Speech Recognition Training
Data” (LB-ASRTD), we trained our ASR model using an
RNN-based end-to-end speech recognition architecture called
“Deep Speech 2” (DS2) [19], developed by Baidu Research
Community. HPC techniques are used extensively in DS2,
resulting in 7x faster training than the previous edition, “Deep
Speech.” Batch normalization is combined with RNN in this
architecture, and a unique optimization curriculum called
SortaGrad [19] is employed.
∞
X
L(x, y; ) = log ΠP ctc(lt|x; θ)
l∈=Align(x,y)
Here, Align(x,y) denotes the set of all possible combinations
of the characters of the transcription y to frames of input x.
C. Sentence summarizing:
Customers’ responses are usually very long which contain
unnecessary information. Our system uses the Bangla Sentence
Summarizing technique to reduce sentence length into the
important point without losing necessary information. So we
used the Seq2Seq Bangla news summarization technique with
attention model to summarize the response of the customer
[26]. Seq2seq model reduces the training loss up to 0.001,
which gives a satisfactory result in our experimental test. Fig-
3 [26] illustrates the sentence summarizing architecture.
1) Data Set and Data Processing : The text data comes
from popular Bangla newspapers “The Daily Prothom Alo”
and “The Daily Ittefaq.” We have selected 250 long ar-
ticles from “Prothom Alo,”, and 250 articles from “The
Daily Ittefaq”. Moreover, BANS [16] dataset is used from
“kaggale.com,” which contains 19k long articles with short
summaries that are used for the model’s evaluation. Collected
data has been processed in four steps; break down sentences
into series of tokens, add the full meaning of the contractions
and remove stopwords and punctuation, apply lemmatization
to convert the word into their root form, and finally, break
words into parts of speech.
2) Count Vocabulary: Vocabulary is counted from the own
dataset among 12,676 unique words are found from the
dataset. Among this unique model only 11,712 words are used
by removing English words from the Bangla Article.
3) Model Architecture: For training, the dataset RNN
encoder-decoder and Seq2Seq learning with attention model
mechanism are used to summarise the articles. This frame-
work has three components: an encoder network, an attention
network, and a decoder network. The encoder converts the
sequence into fixed-size context vector, which represents a
semantic summary of the article. This context vector works
as the initial stage of the decoder connected to hidden units,
although the encoder time stamps are not equal in encoder
and decoder. If A is a long sequence of input where a is
the target sequence of sentences b is the source sequence of
Figure 4: WaveNet Architecture
the sentences, the maximum probability of the word vector
sequence will be
arg(maxbp(a|b)) S2ORC pairsdataset. BERT makes it unsuitable for semantic
similarity search as well as for unsupervised tasks like
Sequence to Sequence model with Bahdanau attention model
clustering. In N. Reimers Al. [17] present Sentence-BERT
is used with fixed output that maps the fixed-length output. A
(SBERT) that use siamese and triplet object function to derive
pre-trained Bangla word vector file “bn w2c model” is used for
semantically meaningful sentence embedding from comparing
word embedding that converts all the words into a numerical
using cosine-similarity. This reduces the effort for finding
form. The final output of the word is in vector form used for
the most similar pair from 65 hours with BERT / Roberta to
model training.
about 5 seconds with SBERT while maintaining the accuracy
D. Workflow of Interactive Agent: from BERT. Given an anchor sentence a, a positive sentence
p, and a negative sentence n, triplet loss tunes the network
We encoded all the sentences from our database using such that the distance between a and p is smaller than the
Sentence Transformer and saved the encoded score in an distance between a and n. Mathematically, we minimize the
array as repeatedly encoding cost us time for every user. The following loss function [17]:
sentences go to the pooling layer, followed by the BERT
model to make a vector to predict a score. With the same max(||sa − sp|| − ||sa − sn||
procedures, we use “Paraphrase mpnet base v2” to encode
With sx, the sentence embedding for a/n/p a distance metric
new sentences that come from users simultaneously. Then
and margin.
new sentence and database sentences array are compared
by similarity check using cosine similarity technique. The E. Text to Speech Synthesis
maximum score is likely the expected query the user is Our intended model can assist customers by interacting with
looking for. The system will transfer the answer via the TTS them; it can reply to various speech in audio format. As we
system, and at the same time, it enlists necessary data, for targeted the domain of e-commerce in Bangladesh, most of
example, doctors’ appointments, into the database. To avoid the users use Bangali Language to communicate. We used
irreverent queries, we set up a thresh holding that is 0.96; the python-based “gTTS” library to turn our text into speech.
this gives a relevant result in our case. The library serves to convert the text by calling API, which
Sentence Transformer: Text from the ASR system in serves the best quality of sound among other available TTS. It
Bangla language, the model encodes every word with uses DeepMind’s WaveNet [15] to deliver the highest accuracy
BERT sentence transformers model using a pertained possible. The architecture of the WaveNet model is shown in
weight called “paraphrase mpnet base v2” [17] has Fig-4 [14]. Furthermore, the speech generation speed from text
been used to convert the word into a vector format. response is quite impressive and useful for real-time uses.
Paraphrase npnet base v2 model is based on Microsoft
mpnet base having a dimension of 768 and mean pooling F. Database
structure. The model is trained with AllNLI, sentence The system consists of three types of Databases, 1. Personal
compression, SimpleWiki, quora duplicates, coco captions, Information Database (PID), 2. Generic Information Database
flickr30k captions,yahoo answers title questions, (GID), 3. Credential Information Database (CID). PID mainly

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 Figure 5: Personal Information Database (PID) Figure 7: Credential Information Database (CID)

Figure 6: Generic Information Database (GID)

receives the user’s name, National Identity (NID) number,

mobile number, date of birth, recorder response, speech
adapted GMM based model and calling time for each user. For Figure 8: Accuracy of Speaker Recognition Module in two
verification, the system matches the personal information given different environments
by the users with the PID’s information. Fig. 5 illustrates the
sample of PID. GID consists of the frequently asked questions
(FAQs) of the e-commerce domain. When a user is new to into the neural network. We used 10 Gated Recurrent Unit
the system, he/she cannot access the CID, where the user’s (GRU) layers with 800 hidden nodes each. The learning rate
confidential information is stored. Instead, they get services was set to 3 x 10-4, and the momentum was set at 0.9.
from the GID. Fig. 6 represents the samples of GID, and Fig. 7 To minimize over-fitting, our ASR model was trained for 30
illustrates the structure of the credential information database. epochs. The LB-ASRTD corpus, which comprises almost 210
hours of utterances, is now the largest open-source speech
III. R ESULT corpus in Bangla. The entire dataset is divided into three sets:
train, validation, and test, each of which contains 171 hours,
To evaluate the speaker recognition module of our proposed 18 hours, and 18 hours of speeches, respectively. Our trained
system, we considered two different environments for testing, model got the word error rate (WER) of 42.15%. Fig. 9 shows
Env 1: Noisy, Env 2: Studio quality. The microphones used the curve of Epoch vs WER for training the Deep Speech 2
for those two environments are also different. At first, for Model.
a noisy environment, We conducted our test with a total
of 42 samples allocated in 7 consecutive steps for each
environment. Afterward, we observed the flow of the changing
accuracy with the increasing number of sample sizes. The
same procedures had been continued for the studio-quality
environment, and then we observed the diversity of accuracy in
different environments. For instance, we can see from Figure
8, when the sample size is only 7, the module recognized all
the 7 persons in both environments, but when we increased the
sample size to 14, the module failed to recognize 1 person in
a noisy environment that reduced the accuracy of the module
to 92.85%. With the increasing sample size, the performance Figure 9: Epoch vs WRE for training Deep Speech 2 Model
tends to be constant after a particular point. We got 94.38%
accuracy in a real-time environment. However, the module’s For training Sentence Summerization model we use
performance might decrease if we test it in large-scale real- , RNN size = 256, batch size = 2, learning rate =
time environments. 0.001, and probability rate = 0.70 to 0.75 with ”Adam”
We extracted the MFCC features from the utterances for optimizer. The model is tested on Bangali Abstractive
our Deep Speech 2 model and then fed these input features News Summarization(BANS) dataset [16]. Using the main

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authors parameters the average loss is 0.004 which is quite
satisfactory.
We used the weight of the BERT sentence
transformer(paraphrase-mpnet-base-v2) trained on STSb,
DupQ, TwitterP, SciDocs, Clustering dataset having average
accuracy 76.84% shown in Table:I
Table I: Accuracy of sentence transformer “paraphrase-mpnet-
base-v2” for the quality to embedded sentences and to embed-
ded search queries paragraphs
Dataset Accuracy(%)
STSb 86.99
DupQ 87.80
TwitterP 76.05
SciDocs 80.57
Clustering 52.81
IV. C ONCLUSION
Our developed system can communicate with users in the
standardized Bangla language. This whole AI-based archi-
tecture will automate the call center as well as increase the
efficiency and quality of services. Moreover, this model can
reduce the waiting time in the call center and make a positive
response to the customers by providing unbiased responses.
But in the real case, our system may not handle converting
regional forms of speech into text. As we don’t introduce
any noise cancellation module during the phone call, the
accuracy may degrade in a noisy environment. The GMM
based MFCC model is suitable for the small size of the dataset.
But increasing numbers of users may cause degrade of the
accuracy. In the future, we have a plan to replace the model
with sincnet to improve the overall accuracy. However, our
system is a milestone for call center support in the Bangla
language, creating state of art for future development to ensure
customer satisfaction.
V. ACKNOWLEDGEMENT
ICT innovation fund from ICT Division, Ministry of Posts,
Telecommunications and Information Technology, the People’s
Republic of Bangladesh funds this research work and pilot
project. The total technical support is provided by AIMS Lab,
United International University, Bangladesh.
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