# An XGB-Based Reliable Transmission Method in the mMTC Scenarios

Authors: Tongyi Zheng; Lei Ning; Qingsong Ye; Fan Jin
Journal: Security and Communication Networks
 (2021)

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## Abstract

Massive machine-type communications (mMTCs) for Internet of things are being developed thanks to the fifth-generation (5G) wireless systems. Narrowband Internet of things (NB-IoT) is an important communication technology for machine-type communications. It supports many different protocols for communication. The reliability and performance of application layer communication protocols are greatly affected by the retransmission time-out (RTO) algorithm. In order to improve the reliability and performance of machine-type communications, this study proposes a novel RTO algorithm UDP-XGB based on the user datagram protocol (UDP) and NB-IoT. It combines traditional algorithms with machine learning. The simulation results show that real round-trip time (RTT) is close to the RTO, which is obtained by this algorithm, and the reliability and performance of machine-type communications have improved.


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## Body

## 1. Introduction

5G, the fifth generation of mobile communication technology, on the one hand, greatly improves the high-bandwidth mobile Internet service experience of individual users and creates new life and entertainment application scenarios. On the other hand, it conforms to 5G network capabilities such as large bandwidth, massive connection, and low delay, and forms a new generation of information infrastructure with other basic common capabilities, such as artificial intelligence (AI), Internet of things (IoT), cloud computing, big data, and edge computing.IoT is an essential technology in 5G mobile telecommunications and is expected to bring enormous economic growth. Due to the rapid development of IoT, machine-type communications (MTCs) have attracted more and more attention and interest from academia and industry. With the increasing popularity of intelligent transportation, smart cities, etc., it is envisioned that the number of IoT devices will reach 75 billion by 2025, which is much larger than the number of mobile phone users. Massive machine-type communications (mMTCs) have been assigned as one of the three use cases for 5G in response to the massive number of IoT devices online at the same time. Wearable devices collecting and uploading small packets of data are becoming an integral part of MTC. To prevent the high-frequency communication of some devices in mMTC from consuming too much network resources, we need to analyze it from different technical levels based on the existing IoT communication technology. From the perspective of technical architecture, the IoT can be divided into three layers, which are the perception layer, the network layer, and the application layer [1].The perception layer is the foundation of the development and application of the IoT. The sensing layer includes various types of data acquisition devices, such as wearable devices, temperature sensors, and humidity sensors, including the sensor network before data are connected to the gateway. The radio frequency identification (RFID) system is the most widely used sensor system in the IoT. It only needs to scan the corresponding electronic tag to obtain the information of the tagged object. RFID tag recognition depends on collision detection, which will significantly affect the performance of tag recognition [2].The network layer of the IoT builds on the existing mobile communication network and the Internet. The network layer is responsible for the data communication between different devices and the cloud. Network communication depends not only on the infrastructure of the Internet, but also on a layered network protocol stack. As shown in Figure1, it can be divided into application layer, transmission layer, network layer, data link layer, and physical layer from the top to the bottom. Each layer of the protocol is responsible for managing and scheduling a different infrastructure to perform a given task. There are many communication protocols in different protocol stack layers, which can form many different communication modes. The combination of different communication protocols is suitable for different application scenarios under the IoT.Figure 1
Network protocol stack layers.The application layer consists of various data processing and analysis program applications. The application layer is responsible for data mining and analysis of all kinds of data collected by the perception layer and then presents it to users through intuitive data visualization, such as line diagram and scatter diagram. The users use these data to grasp environmental information and make corresponding decisions.The network layer of the IoT needs to specify a network communication technology, which is the physical layer of the protocol stack. Choosing different communication modes at the physical layer of the protocol stack can make a big performance difference. In this study, narrowband Internet of things (NB-IoT) technology, a low-power wide area network technology, is chosen to communicate. NB-IoT is one of the cornerstones of the large-scale IoT technology that forms the basis of 5G. Several features of NB-IoT are shown as follows:(1)
Low cost: the NB-IoT network can be upgraded on the basis of existing long-term evolution (LTE) network, which greatly reduces the cost of network construction and maintenance.(2)
Deep coverage: through time-domain retransmission technology and improved power spectral density, NB-IoT improves the maximum coupling loss (MCL) by 20 dB compared with Global System for Mobile Communications (GSM), covers three times the distance of GSM, and can penetrate two more walls than GSM. MCL is the maximum total channel loss between the device and the antenna port of the base station when transmitting data. The link enhances and the signal coverage expands with the increase in MCL value.(3)
Low power consumption: NB-IoT technology has designed three different power-saving modes. The device can choose the most appropriate power-saving mode according to its own business characteristics to achieve the purpose of minimizing power consumption, achieving real superlong standby, and greatly extending battery life.(4)
Massive connectivity: NB-IoT networks allow more devices to be connected simultaneously, 50 to 100 times faster than existing wireless technologies. According to the simulation test, now the single-cell base station of NB-IoT network can access about 50,000 terminal devices.NB-IoT communication also needs the protocol support of transport layer, such as transmission control protocol (TCP) and user datagram protocol (UDP).TCP provides a connection-oriented, reliable byte stream service. Connection-oriented means that two applications using TCP must first establish a TCP connection before they can exchange packets with each other. The process is similar to making a phone call, waiting until the communication is over before closing the connection. In a TCP connection, only two parties communicate with each other. Broadcast and multicast cannot be used with TCP. TCP uses a serial number and an acknowledgement number to acknowledge receipt of relevant data. The TCP service on the destination host acknowledges the received data and sends the acknowledgement information to the source application. The size of the data that the source host can transmit before receiving the acknowledgement message is called the window size. For the management of lost data and flow control, TCP starts a retransmission timer when sending a piece of data. If no acknowledgement is received before the retransmission timer time-outs, the data segment is retransmitted. TCP is not suitable for NB-IoT round-the-clock data collection and reporting services because TCP needs to maintain network connections.UDP is connectionless, and this means that no connection needs to be established before sending data and no connection can be released at the end of sending data, reducing overhead and delays before sending data. UDP uses best effort delivery; that is, reliable delivery is not guaranteed and the host does not need to maintain a complex list of connection states. UDP has no congestion control, and any congestion that occurs on the network will not slow down the transmission rate of the source host. This is important for some real-time applications. In addition, UDP supports one-to-one, one-to-many, many-to-one, and many-to-many interactive communications. Finally, UDP has a small header overhead of only 8 bytes, which is shorter than TCP’s 20 byte header. So, UDP is more suitable for NB-IoT round-the-clock data collection and reporting services because UDP is lightweight and connectionless [3].Due to the above characteristics, UDP is not reliable in data transmission. In practical applications, it is often necessary to ensure that the data reach the other end. This requires the development of application layer protocols based on UDP, such as constrained application protocol (CoAP), and the addition of a timing retransmission mechanism to ensure reliability. The most important part of the timed retransmission is the determination of the retransmission time-out (RTO). The device will determine whether the data have reached the other end after a certain amount of time based on the RTO. If the data do not reach the other end, the device needs to send the data again. The RTO needs to be determined based on the current state of the network. When a device using NB-IoT is moving, the network will produce large fluctuations, such as from outdoor to indoor. At this time, the determination of RTO is often not ideal, resulting in high network delay of data transmission or large network resource consumption.Based on the above observations, this study studies how to determine the RTO when NB-IoT uses UDP for reliable transmission in mobile scenarios. The objective is how to send the same data with less network resources and lower network delay compared with traditional algorithms.The rest of the paper is arranged as follows. Section2 gathers a literature review made of some related works, while Section 3 shows the description of transport model, traditional algorithm, and target problem. Section 4 describes the UDP-XGB algorithm and its details. In Section 5, simulation tests are presented and analyzed. Eventually, Section 6 highlights conclusions.


## 2. Related Works

There are many communication technologies in the field of IoT. These communication technologies are based on low-power wide area network (LPWAN) technology. To determine which communication technologies are more likely to become mainstream at large-scale IoT in the future, we conducted a comprehensive survey of LPWAN. Firstly, we looked at the development and status of LPWAN [4, 5]. In the IoT communication security, we also found a good way to deal with [6]. Then, we investigated the deployment of different technologies in large-scale IoT [7], and NB-IoT is more suitable for large-scale deployment because it can accommodate massive connections. In addition, we investigate energy consumption analysis and IoT application life of different technologies [8, 9], because NB-IoT technology needs to ensure deeper and wider signal coverage, energy consumption is slightly higher than other technologies. We then surveyed the coverage of the different technologies [10], and NB-IoT technology can cover many harsh environments. Finally, we investigated the NB-IoT technology in depth [11–13]. The technology is a good candidate for large-scale IoT due to its enhanced indoor coverage, delay insensitivity, and support for massive connections.Then, we found a key problem in NB-IoT data transmission optimization and how to effectively determine RTO. Since RTO can significantly affect the performance of the transport protocol [14, 15], a good RTO algorithm is critical. The RTO algorithm was proposed at the beginning of TCP [16]. To improve TCP performance, a variety of RTO algorithms have been proposed [17–20]. At the same time, there is also a UDP-based RTO algorithm [21–23]. In addition, there are also some RTO algorithms for other scenarios [24, 25]. The above algorithms are based on statistical RTT to calculate RTO, and these algorithms are slow to the fluctuations of network signal. When the device is moving, it is easy to switch the scene, resulting in large network fluctuations. So, we need an algorithm that is more sensitive to fluctuations in network signal. Therefore, we want to calculate RTO by network signal. Our idea came from two studies. Kotagi V. J. et al. proposed the breathing method of NB-IoT, which can adjust the transmission power of the equipment through the fluctuations of the network signal [26]. Caso G. et al. predicted the success of random access of NB-IoT and long-term evolution (LTE) networks based on the network status and adjusted the power of random access using the predicted results [27]. We use machine-learning methods to analyze the collected data and propose a UDP-XGB algorithm [28]. This study further improves the performance of the UDP-XGB algorithm, and enriches and improves the experimental simulation.


## 3. Transmission Model and Problem Formulation

In this section, we introduce a simple UDP communication model to simulate data transmission in a real-world scenario. We referred to several UDP transmission model [29–32] and simplified the rest as much as possible. The purpose is to better focus on determining the RTO for timed retransmission. In addition, we also introduce several algorithms for determining the RTO. Finally, we formally define our target problem.


### 3.1. Transmission Model Description

The UDP has two problems that need to be solved. Some real-time applications need to use UDP without congestion control. However, when many source hosts send real-time data streams with high speed to the network at the same time, the network may be congested, causing everyone to be unable to receive normally. On the other hand, some real-time applications that use UDP need to make appropriate improvements to the unreliable transport of UDP to reduce data loss. The application process can add some measures to improve the reliability without affecting the real-time performance of the application, such as retransmitting lost messages. To solve these two problems, most application layer transport protocols based on UDP use sequential transmission and timed retransmission to solve them. UDP transmission model is shown in Figure2. It mainly contains the following two functions.(i)
Sequential Transmission: it specifies a message queue of length N. Each time a message is sent, the ID is bound to identify the order in which the message is sent. After each message is sent, the message queue stores the corresponding ID and the queue length is incremented by one. After receiving an acknowledge character (ACK) from the server, the corresponding ID in the message queue is removed and the queue length is reduced by one. The message is sent at a fixed time interval T, but when the message queue length is N, that is, there are N messages to be acknowledged, the client stops sending messages and waits for an ACK from the server for the last message until the message queue length is less than N.(ii)
Timed Retransmission: after each message is sent, a timer will be set. When the timer exceeds the specified RTO and does not receive an ACK from the server about the corresponding message ID, the message will be sent again.Figure 2
UDP transmission model.The time between the message being sent and the corresponding ACK being received is called round-trip time (RTT). If the RTO is less than the RTT, clients will recognize the message loss before receiving the corresponding ACK and will send the message again, resulting in many unnecessary retransmissions. If the RTO is much larger than the RTT, the message cannot be sent again in time when the message is sent and the ACK is lost, resulting in a certain transmission delay in the message transmission. To take into account both network resource consumption and transmission delay, RTO should be slightly larger than RTT.When timed retransmission occurs, it is difficult to accurately count the RTT. In this situation, general practice is not to count the RTT of timed retransmission. RTT of timed retransmission is also not counted when RTT is counted in this study.Depending on the relationship between RTT and RTO, the data are sent in three different states. When RTT is 0, the data sent fails. RTT> RTO represents false retransmission. RTT <= RTO indicates that the data have been sent successfully.


### 3.2. Traditional Algorithm Description

There are many different ways to determine the RTO. Standard TCP algorithm [16] and CoAP-Eifel algorithm [22] are used to determine RTO by historical RTT. The RTO is determined by random value in CoAP [21]. The RTO will double when data retransmission occurs in these algorithms.


### 3.3. Standard TCP

(1)SRTT=1−α⋅SRTT+α⋅RTT.SRTT represents the mean value of recent RTT. α is a smoothing factor. As the α decreases, the SRTT becomes more stable and the SRTT is less affected by the current RTT.(2)δ=RTT−SRTT.δ represents the error between the SRTT and the measured RTT. It shows the fluctuation of RTT from SRTT.(3)RTTVAR=1−β⋅RTTVAR+β⋅δ.RTTVAR represents the recent average deviation between SRTT and RTT. The absolute error δ represents the current deviation.(4)RTO=SRTT+maxG,K⋅RTTVAR.The new RTO is obtained bySRTT and RTTVAR, and the recommended value of K is 4, and G represents the minimum time interval of the timer.(5)RTO=RTO⋅2.When data packets are lost, the newRTO is twice that of the previous RTO.(6)SRTT=RRTTVAR=R2RTO=SRTT+maxG,K⋅RTTVAR.When the first RTT measurementR is made, the host should be set as above.


### 3.4. CoAP-Eifel

(7)δ=RTT−SRTT.δ represents the error between the SRTT and the measured RTT. It shows the fluctuation of RTT from SRTT.(8)GAIN=13.GAIN represents the rate of change in SRTT, which is taken as 1/3 in this study.(9)GAIN¯=GAIN,δ−RTTVAR≥0,GAIN2,δ−RTTVAR<0..GAIN¯ is the rate of change in RTTVAR. δ−RTTVAR≥0 indicates that there is a large error between the previously estimated RTT and real RTT, and GAIN¯ remains unchanged. δ−RTTVAR<0 indicates that the error between the previous estimated RTT and real RTT is small or the estimated RTT is greater than RTT, so GAIN¯ is appropriately reduced to maintain the stability of RTTVAR.(10)SRTT=SRTT+GAIN⋅δ.SRTT represents the mean value of recent RTT. SRTT consists of the previous SRTT and the estimated error δ. The influence of estimation error δ on SRTT increases with the increase in GAIN. SRTT produces large fluctuations, but they also quickly correct and follow the RTT fluctuations when the RTT fluctuates.(11)RTTVAR=RTTVAR+GAIN¯⋅δ−RTTVAR,δ≥0RTTVAR,δ<0.RTTVAR represents the recent average deviation between SRTT and RTT. RTTVAR consists of the previous RTTVAR and the estimated error δ. δ≥0 means that the real RTT is higher than the mean value of the past RTT, and the current estimated RTT is likely to be less than the real RTT, so the estimated RTT needs to be revised. δ<0 means that the real RTT is lower than the mean value of the past RTT, and the current estimated RTT has a good effect, so there is no need to revise the estimated RTT.(12)RTO=maxSRTT+RTTVARGAIN,RTT+G.The RTO takes the maximum value between the estimated RTT—SRTT+RTTVAR/GAIN and the last real RTT. The last real RTT can be used as an estimate of the current RTT because the difference between the real RTT values is often very small during successive transmission. G represents the minimum time interval of the timer.(13)RTO=RTO⋅2.When data packets are lost, the newRTO is twice that of the previous RTO.


### 3.5. CoAP

(14)RTO=TIMEOUT⋅γ,where TIMEOUT is the basic time-out value, and the typical value is 2000 ms; γ is a time-out random wavy factor, and it is random value between 1.0 and 1.5 in general; and RTO is a fluctuant value based on TIMEOUT and γ.(15)TIMEOUTM=TIMEOUT⋅γ⋅RETRANSM,where RETRANSM is the max number of retransmission, and the typical value is 4; TIMEOUTM is max time of RTO.(16)RTO=minRTO⋅2,TIMEOUTM.When data packets are lost, the newRTO is twice that of the previous RTO, but it should be less than TIMEOUTM.The variables used in the traditional algorithm description are summarized in Table1.Table 1
Summary of algorithm variables.
SymbolDescriptionRTTThe time between the message being sent and the corresponding ACK being received.SRTTThe approximate mean value of historical statistics RTT.αThe smooth factor to SRTT. It is 1/8 on this study.RTOThe time interval at which data are sent again.βThe smooth factor toRTTVAR. It is 1/4 on this study.RFirst RTT measurement value.δThe error between the calculated SRTT of the last data transmission and the real measured RTT.RTTVARThe deviation correction between SRTT and real RTT.KA fixed coefficient in standard TCP is 4.GThe minimum time interval of the timer. It is 100 ms on this study.GAINThe rate of change in SRTT is 1/3.TIMEOUTThe basic time-out value is 2000 ms in general.γThe random value is between 1.0 and 1.5 in general.RETRANSMThe max number of retransmission is 4 in general.TIMEOUTMThe max time of RTO.GAIN¯The rate of change inRTTVAR.


### 3.6. Problem Formulation

The state of the network can affect the speed of data transmission. When the network is in a good state, data tend to get to the other end more quickly. In the case of poor network signal, data may be lost during transmission. Based on this observation, it is assumed that network signal affects RTT. We need to find out how the network signal affects the RTT, so that we can use the network signal to estimate the possible RTT for this data transmission. UDP consumes less network resources and has lower transmission delays while ensuring reliability using the estimated RTT as the RTO.Input Instance: network status indicator data are collected by NB-IoT terminals, such as reference signal receiving quality (RSRQ), reference signal receiving power (RSRP), signal–to- interference-plus-noise ratio (SINR), and received signal strength indication (RSSI). In addition, the real RTT is counted when collecting network status data.Output Instance: network signal data and real RTT are observed and analyzed to find the implicit quantitative relationship between network signal data and real RTT.Objective: the quantitative relationship between the obtained network status data and the real RTT is used as the UDP-XGB algorithm. UDP using the UDP-XGB algorithm can reduce some unnecessary retransmission.Packet loss rate: the packet loss rate is the ratio of the number of packets lost to the group of data sent during the test.(17)PLR=SfailStotle.PLR represents packet loss rate. Stotle represents the total number of packages sent in the test. Sfail represents the total number of packages ended, when RTT is 0 or RTT>RTO. PLR mainly shows the proportion of the amount of resent data due to sent fails to the amount of the task data. The lower the packet loss rate, the higher the probability that the data will be sent once and the less network resources will be consumed by sending the same amount of data.The variables used in the problems are summarized in Table2.Table 2
Summary of variables.
SymbolDescriptionPLRThe packet loss rate is the ratio of the number of packets lost to the group of data sent during the test.StotleThe total number of packages sent for this task.SfailThe number of packets sent when false retransmission or transmission is failed.


## 3.1. Transmission Model Description

The UDP has two problems that need to be solved. Some real-time applications need to use UDP without congestion control. However, when many source hosts send real-time data streams with high speed to the network at the same time, the network may be congested, causing everyone to be unable to receive normally. On the other hand, some real-time applications that use UDP need to make appropriate improvements to the unreliable transport of UDP to reduce data loss. The application process can add some measures to improve the reliability without affecting the real-time performance of the application, such as retransmitting lost messages. To solve these two problems, most application layer transport protocols based on UDP use sequential transmission and timed retransmission to solve them. UDP transmission model is shown in Figure2. It mainly contains the following two functions.(i)
Sequential Transmission: it specifies a message queue of length N. Each time a message is sent, the ID is bound to identify the order in which the message is sent. After each message is sent, the message queue stores the corresponding ID and the queue length is incremented by one. After receiving an acknowledge character (ACK) from the server, the corresponding ID in the message queue is removed and the queue length is reduced by one. The message is sent at a fixed time interval T, but when the message queue length is N, that is, there are N messages to be acknowledged, the client stops sending messages and waits for an ACK from the server for the last message until the message queue length is less than N.(ii)
Timed Retransmission: after each message is sent, a timer will be set. When the timer exceeds the specified RTO and does not receive an ACK from the server about the corresponding message ID, the message will be sent again.Figure 2
UDP transmission model.The time between the message being sent and the corresponding ACK being received is called round-trip time (RTT). If the RTO is less than the RTT, clients will recognize the message loss before receiving the corresponding ACK and will send the message again, resulting in many unnecessary retransmissions. If the RTO is much larger than the RTT, the message cannot be sent again in time when the message is sent and the ACK is lost, resulting in a certain transmission delay in the message transmission. To take into account both network resource consumption and transmission delay, RTO should be slightly larger than RTT.When timed retransmission occurs, it is difficult to accurately count the RTT. In this situation, general practice is not to count the RTT of timed retransmission. RTT of timed retransmission is also not counted when RTT is counted in this study.Depending on the relationship between RTT and RTO, the data are sent in three different states. When RTT is 0, the data sent fails. RTT> RTO represents false retransmission. RTT <= RTO indicates that the data have been sent successfully.


## 3.2. Traditional Algorithm Description

There are many different ways to determine the RTO. Standard TCP algorithm [16] and CoAP-Eifel algorithm [22] are used to determine RTO by historical RTT. The RTO is determined by random value in CoAP [21]. The RTO will double when data retransmission occurs in these algorithms.


## 3.3. Standard TCP

(1)SRTT=1−α⋅SRTT+α⋅RTT.SRTT represents the mean value of recent RTT. α is a smoothing factor. As the α decreases, the SRTT becomes more stable and the SRTT is less affected by the current RTT.(2)δ=RTT−SRTT.δ represents the error between the SRTT and the measured RTT. It shows the fluctuation of RTT from SRTT.(3)RTTVAR=1−β⋅RTTVAR+β⋅δ.RTTVAR represents the recent average deviation between SRTT and RTT. The absolute error δ represents the current deviation.(4)RTO=SRTT+maxG,K⋅RTTVAR.The new RTO is obtained bySRTT and RTTVAR, and the recommended value of K is 4, and G represents the minimum time interval of the timer.(5)RTO=RTO⋅2.When data packets are lost, the newRTO is twice that of the previous RTO.(6)SRTT=RRTTVAR=R2RTO=SRTT+maxG,K⋅RTTVAR.When the first RTT measurementR is made, the host should be set as above.


## 3.4. CoAP-Eifel

(7)δ=RTT−SRTT.δ represents the error between the SRTT and the measured RTT. It shows the fluctuation of RTT from SRTT.(8)GAIN=13.GAIN represents the rate of change in SRTT, which is taken as 1/3 in this study.(9)GAIN¯=GAIN,δ−RTTVAR≥0,GAIN2,δ−RTTVAR<0..GAIN¯ is the rate of change in RTTVAR. δ−RTTVAR≥0 indicates that there is a large error between the previously estimated RTT and real RTT, and GAIN¯ remains unchanged. δ−RTTVAR<0 indicates that the error between the previous estimated RTT and real RTT is small or the estimated RTT is greater than RTT, so GAIN¯ is appropriately reduced to maintain the stability of RTTVAR.(10)SRTT=SRTT+GAIN⋅δ.SRTT represents the mean value of recent RTT. SRTT consists of the previous SRTT and the estimated error δ. The influence of estimation error δ on SRTT increases with the increase in GAIN. SRTT produces large fluctuations, but they also quickly correct and follow the RTT fluctuations when the RTT fluctuates.(11)RTTVAR=RTTVAR+GAIN¯⋅δ−RTTVAR,δ≥0RTTVAR,δ<0.RTTVAR represents the recent average deviation between SRTT and RTT. RTTVAR consists of the previous RTTVAR and the estimated error δ. δ≥0 means that the real RTT is higher than the mean value of the past RTT, and the current estimated RTT is likely to be less than the real RTT, so the estimated RTT needs to be revised. δ<0 means that the real RTT is lower than the mean value of the past RTT, and the current estimated RTT has a good effect, so there is no need to revise the estimated RTT.(12)RTO=maxSRTT+RTTVARGAIN,RTT+G.The RTO takes the maximum value between the estimated RTT—SRTT+RTTVAR/GAIN and the last real RTT. The last real RTT can be used as an estimate of the current RTT because the difference between the real RTT values is often very small during successive transmission. G represents the minimum time interval of the timer.(13)RTO=RTO⋅2.When data packets are lost, the newRTO is twice that of the previous RTO.


## 3.5. CoAP

(14)RTO=TIMEOUT⋅γ,where TIMEOUT is the basic time-out value, and the typical value is 2000 ms; γ is a time-out random wavy factor, and it is random value between 1.0 and 1.5 in general; and RTO is a fluctuant value based on TIMEOUT and γ.(15)TIMEOUTM=TIMEOUT⋅γ⋅RETRANSM,where RETRANSM is the max number of retransmission, and the typical value is 4; TIMEOUTM is max time of RTO.(16)RTO=minRTO⋅2,TIMEOUTM.When data packets are lost, the newRTO is twice that of the previous RTO, but it should be less than TIMEOUTM.The variables used in the traditional algorithm description are summarized in Table1.Table 1
Summary of algorithm variables.
SymbolDescriptionRTTThe time between the message being sent and the corresponding ACK being received.SRTTThe approximate mean value of historical statistics RTT.αThe smooth factor to SRTT. It is 1/8 on this study.RTOThe time interval at which data are sent again.βThe smooth factor toRTTVAR. It is 1/4 on this study.RFirst RTT measurement value.δThe error between the calculated SRTT of the last data transmission and the real measured RTT.RTTVARThe deviation correction between SRTT and real RTT.KA fixed coefficient in standard TCP is 4.GThe minimum time interval of the timer. It is 100 ms on this study.GAINThe rate of change in SRTT is 1/3.TIMEOUTThe basic time-out value is 2000 ms in general.γThe random value is between 1.0 and 1.5 in general.RETRANSMThe max number of retransmission is 4 in general.TIMEOUTMThe max time of RTO.GAIN¯The rate of change inRTTVAR.


## 3.6. Problem Formulation

The state of the network can affect the speed of data transmission. When the network is in a good state, data tend to get to the other end more quickly. In the case of poor network signal, data may be lost during transmission. Based on this observation, it is assumed that network signal affects RTT. We need to find out how the network signal affects the RTT, so that we can use the network signal to estimate the possible RTT for this data transmission. UDP consumes less network resources and has lower transmission delays while ensuring reliability using the estimated RTT as the RTO.Input Instance: network status indicator data are collected by NB-IoT terminals, such as reference signal receiving quality (RSRQ), reference signal receiving power (RSRP), signal–to- interference-plus-noise ratio (SINR), and received signal strength indication (RSSI). In addition, the real RTT is counted when collecting network status data.Output Instance: network signal data and real RTT are observed and analyzed to find the implicit quantitative relationship between network signal data and real RTT.Objective: the quantitative relationship between the obtained network status data and the real RTT is used as the UDP-XGB algorithm. UDP using the UDP-XGB algorithm can reduce some unnecessary retransmission.Packet loss rate: the packet loss rate is the ratio of the number of packets lost to the group of data sent during the test.(17)PLR=SfailStotle.PLR represents packet loss rate. Stotle represents the total number of packages sent in the test. Sfail represents the total number of packages ended, when RTT is 0 or RTT>RTO. PLR mainly shows the proportion of the amount of resent data due to sent fails to the amount of the task data. The lower the packet loss rate, the higher the probability that the data will be sent once and the less network resources will be consumed by sending the same amount of data.The variables used in the problems are summarized in Table2.Table 2
Summary of variables.
SymbolDescriptionPLRThe packet loss rate is the ratio of the number of packets lost to the group of data sent during the test.StotleThe total number of packages sent for this task.SfailThe number of packets sent when false retransmission or transmission is failed.


## 4. Proposed UDP-XGB

To better adapt to the large RTT fluctuations caused by scene switching during the movement of NB-IoT, such as from outdoor to indoor, this study uses the machine-learning (ML) method to forecast the RTT and take the predicted RTT as the RTO.In this study, we collected four kinds of network signal features, such as RSRQ, RSRP, SINR, and RSSI. Then, the Pearson correlation coefficient method is used to analyze the characteristics of the acquired network signals. As shown in Figure3, these network signals have some correlations with RTT. We proposed a UDP-XGB algorithm based on the four network signals and machine learning.Figure 3
Network signals and RTT Pearson’s correlation coefficient.Extreme gradient boosting (XGBoost) is an algorithm or engineering implementation based on gradient boosting decision tree (GBDT) [33]. In this study, the data of the above four dimensions were used as the characteristic input. RTT was used as the target output. Root-mean-square error (RMSE) was selected as the loss function. 150 regression trees were trained and integrated into a model. The model starts with only one regression tree, and each iteration will find and integrate a new regression tree, which needs to satisfy the target function. Reference (18) is the target function of Algorithm 1. The characteristics of all the data are input into the model and the RTT is predicted as the RTO. The UDP-XGB algorithm in this study is shown in Algorithm 1.(18)Lϕ=RMSERTT,RTT^+∑t=1kΩft,RMSERTT,RTT^=∑i=1nRTT−RTT^2n,Ωft=γ⋅T+12λ⋅∑j=1Tωj2.RMSERTT,RTT^ is loss function. Ωft is a regular item. RTT is a real RTT. RTT^ is the RTT predicted by the current model. n is the number of data samples. ft is a regression tree, a function of input mapped to output. k is the number of regression trees integrated by the current model. T is the number of leaves in the regression tree ft. ωj2 is the square of the score of the leaves in the regression tree ft. γ and λ are the hyperparameters used to prevent overfitting (Algorithm 1).Algorithm 1: UDP-XGB algorithm.
Input: The four features with the highest correlation, RSRQ, RSRP, SINR, and RSSI, are taken as input X. RTT is the target output Y. RTT^ is the RTT predicted by the model. ft is a regression tree, a function of X mapped to Y. Set the target function to Lϕ. Set the number of trees for model integration to num  = 150. RTTVAR represents the deviation between RTT^ and real RTT. β is a smooth factor for RTTVAR, which is 0.25.Output: Model is the mapping relationship between features and RTT, which integrated all of ft. Use the model to predict RTT. RTO is obtained by RTT^ and RTTVAR.(1)
Model is empty(2)
fort = 1 tonumdo(3)
Divide byX to find all the regression trees(4)
Choose a treeft to satisfy minLϕ(5)
Model addft(6)
end for(7)
while send data do(8)
if Retransmit then(9)
RTO=2⋅RTO(10)
else(11)
RTT^  = Model Predict(X)(12)
RTO=RTT^+4⋅RTTVAR(13)
end if(14)
RTO=minRTT^⋅4,RTO(15)
if Send SUCCESS then(16)
RTTVAR=1−β⋅RTTVAR+β⋅RTT−RTT^(17)
end if(18)
end whileUDP-XGB needs to input some data to train a model and to predict RTT by this model. In this study, we input 17 000 data to train models. The function ofRTT^ is similar to SRTT in traditional algorithms. We analyzed 17,000 data collected and found that the fluctuation range of RTT was always less than 4 times of SRTT, so we set the upper limit of RTO as 4 times of RTT^. RTO obtained by predicted RTT adds current deviation RTTVAR between predicted RTT and real RTT. Therefore, RTO can always be greater than the real RTT no matter how the real RTT varies. In addition, when data retransmit due to sent fails, we ignore RTTVAR and the new RTO is twice that of previous RTO.


## 5. Simulation Results

The simulation model in this study adopts static data simulation analysis and carries out simulation analysis among four groups of different algorithms on the collected 2000 network signal to ensure the fairness of simulation data among different algorithms. Assume each data as a round sent.The first round of real RTT is the initialRTT and RTO of all algorithms. The initial SRTT is 1/2 of initial RTT. The initial RTTVAR is 1/4 of initial RTT. We input RTO, SRTT, and RTTVAR into standard TCP, CoAP-Eifel, and CoAP to obtain the RTO, SRTT, and RTTVAR of the next round. When RTT is 0 or RTO is smaller than RTT, packets are lost. If packets are lost, the RTO is calculated using the exponential rollback method. We counted 2000 round RTT and RTO. The packet loss rate and transmission delay were obtained according to the statistical RTO and RTT. In UDP-XGB, we input 17 000 training data into XGBoost to train the model, which can get RTT^. RSRP, RSRQ, RSSI, and SINR in the test data are input into XGBoost model to get RTT^ each round. RTT^ combined with RTTVAR to produce RTO and new RTTVAR. The statistics of packet loss rate and transmission delay are consistent with other algorithms.Figure4 shows the RTO from standard TCP compared with the real RTT. It is clear that RTO can wave in response to fluctuations in the real RTT, but the RTO wave is always behind the RTT wave, so that data will retransmit when RTT has extreme waves. As a result of standard TCP, RTO base is calculated in historical RTT, and the RTO has hysteretic nature and lacks some timeliness. In addition, it is easy to packet loss when network acutely fluctuates due to the hysteresis nature of standard TCP.Figure 4
Comparison of RTT and RTO fluctuations for standard TCP.Figure5 shows the RTO from CoAP-Eifel compared with the real RTT. CoAP-Eifel is similar to standard TCP in that the RTO they produce can wave with fluctuations in the real RTT, but its RTO has hysteretic nature the same as standard TCP. In addition, CoAP-Eifel has greater changes in the face of network signal fluctuations, which can avoid some appearance of false retransmission. However, the cost of acute RTO fluctuations is a long wait when a sent fails, which can significantly affect network transmission performance.Figure 5
Comparison of RTT and RTO fluctuations for CoAP-Eifel.Figure6 shows the RTO from CoAP compared with the real RTT. CoAP is different from the above two methods because its RTO is random and does not fluctuate with the fluctuation of RTT. This means that you need to carefully configure parameters for different network environments, so this approach is difficult to apply to a wide range of networks.Figure 6
Comparison of RTT and RTO fluctuations for CoAP.Figure7 shows the RTO from UDP-XGB compared with the real RTT. UDP-XGB can obtain a base RTT in advance based on network signals, which ensures the timeliness of the RTO at this moment. As RTT surges in the figure, RTO also increases correspondingly. UDP-XGB uses the base RTT that adds the recent predicted deviation to obtain the RTO, so that the RTT is still valid even if real RTT has some fluctuation. Compared with the above three algorithms, UDP-XGB has a more stable RTO.Figure 7
Comparison of RTT and RTO fluctuations for UDP-XGB.Figure8 shows a box diagram of the error distribution between the RTO obtained by different algorithms and the real RTT. The yellow part shows the distribution range of the major errors. The red dots indicate extreme errors. As is shown in Figure 8, CoAP-Eifel error is large and widely distributed. Compared with other algorithms, UDP-XGB has smaller extreme error, which indicates that the UDP-XGB algorithm has higher stability and accuracy in the same network.Figure 8
Error distribution between RTT and RTO.Figure9 shows the packet loss rate of simulation with different algorithms. The packet loss rate of all algorithms is less than 0.1, which indicates that all algorithms have high reliability. Standard TCP has the highest packet loss rate due to its hysteretic nature. CoAP-Eifel has the lowest packet loss rate thanks to its RTO adopting a more drastic increase in network fluctuation, but this also reduces some network transmission performance. The packet loss rate of UDP-XGB is also at a good level compared with other algorithms. This is due to its ability to obtain base RTT in advance according to network signals and avoid some packet loss caused by network fluctuation.Figure 9
Packet loss rate of four algorithms.Figure10 shows the transmission delay of simulation with different algorithms. Transmission delay is the time it takes for a packet to arrive from one end to the other. In this simulation, RTT in the case of successful transmission and RTO in the case of failed transmission are regarded as the transmission delay of each data transmission. It can be seen from Figure 10 that the transmission delay of UDP-XGB algorithm is significantly lower than that of other algorithms, because UDP-XGB limits the upper limit of RTO, thus reducing the waiting time in the case of data packet loss. Because other algorithms do not limit the upper limit of RTO, RTO will quickly grow to a huge value in the case of frequent packet loss. This creates large wait times and transmission delays. This is clearly unreasonable.Figure 10
Transmission delay of four algorithms.


## 6. Conclusion

UDP-XGB algorithm is proposed in this study, which uses UDP to carry out reliability transmission in mobile scenarios for 5G NB-IoT. The three traditional algorithms are compared with UDP-XGB, and the simulation results show that UDP-XGB performs well in packet loss rate and transmission delay. As shown in Figure8, the RTO of UDP-XGB between real RTT values has some deviations, and we need to optimize algorithm to acquire more accuracy. However, UDP-XGB can be applied to other network data transmissions due to its reliability and stability.



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*Source: 9929051-2021-12-26.xml*