Face angle
The authors representing this approach are Brown and Levinson (1978/1987). The names of Brown and Levinson are almost synonymous with the word “politeness” as one researcher said “it is impossible to talk about politeness without mentioning Brown and Levinson” [111; p. 11]. Like Lakoff, Brown and Levinson see politeness as the avoidance of conflict in communication, but their explanatory tools are different from Lakoff’s. The central issues in Brown and Levinson’s theory are “rationality” and “face”, both of which are considered to have universal characteristics. According to the two authors, face is a very sensitive factor, can be damaged, can be maintained and enhanced. Politeness is a system of strategies to mitigate face-threatening acts. Brown and Levinson proposed five strategies of linguistic interaction and asserted that this strategy model has universal application. In communication, the speaker must calculate and consider the level of face threat of the speech act he intends to perform in order to find ways to reduce the level of face threat.
Collaborative conversation perspective
B. Fraser (1975), B. Fraser and Nolen (1981) are the authors representing this trend. According to them, “politeness is not as Lakoff and Leech said, making the listener feel happy and comfortable, nor as Brown and Levinson understood, making the listener not feel uncomfortable, but simply, with an assigned task, we must complete it in the light of the principle of conversational cooperation”. [17; p. 57]. Thus, politeness is controlled by conversational rules and conversational cooperation. When participants comply with those rules, they have achieved the requirement of politeness. On the contrary, when they act incorrectly, they will be considered impolite.
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Brent Davies, Linda Ellison & Christopher Bowring - Carr, School Leadership In The 21st Century; Developing A Strategic Approach, London New York Routledgefalmer 2005 - 87. -
Qos Assurance Methods for Multimedia Communications
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low. The EF PHB requires a sufficiently large number of output ports to provide low delay, low loss, and low jitter.
EF PHBs can be implemented if the output port's bandwidth is sufficiently large, combined with small buffer sizes and other network resources dedicated to EF packets, to allow the router's service rate for EF packets on an output port to exceed the arrival rate λ of packets at that port.
This means that packets with PHB EF are considered with a pre-allocated amount of output bandwidth and a priority that ensures minimum loss, minimum delay and minimum jitter before being put into operation.
PHB EF is suitable for channel simulation, leased line simulation, and real-time services such as voice, video without compromising on high loss, delay and jitter values.
Figure 2.10 Example of EF installation
Figure 2.10 shows an example of an EF PHB implementation. This is a simple priority queue scheduling technique. At the edges of the DS domain, EF packet traffic is prioritized according to the values agreed upon by the SLA. The EF queue in the figure needs to output packets at a rate higher than the packet arrival rate λ. To provide an EF PHB over an end-to-end DS domain, bandwidth at the output ports of the core routers needs to be allocated in advance to ensure the requirement μ > λ. This can be done by a pre-configured provisioning process. In the figure, EF packets are placed in the priority queue (the upper queue). With such a length, the queue can operate with μ > λ.
Since EF was primarily used for real-time services such as voice and video, and since real-time services use UDP instead of TCP, RED is generally
not suitable for EF queues because applications using UDP will not respond to random packet drop and RED will strip unnecessary packets.
2.2.4.2 Assured Forwarding (AF) PHB
PHB AF is defined by RFC 2597. The purpose of PHB AF is to deliver packets reliably and therefore delay and jitter are considered less important than packet loss. PHB AF is suitable for non-real-time services such as applications using TCP. PHB AF first defines four classes: AF1, AF2, AF3, AF4. For each of these AF classes, packets are then classified into three subclasses with three distinct priority levels.
Table 2.8 shows the four AF classes and 12 AF subclasses and the DSCP values for the 12 AF subclasses defined by RFC 2597. RFC 2597 also allows for more than three separate priority levels to be added for internal use. However, these separate priority levels will only have internal significance.
PHB Class
PHB Subclass
Package type
DSCP
AF4
AF41
Short
100010
AF42
Medium
100100
AF43
High
100110
AF3
AF31
Short
011010
AF32
Medium
011100
AF33
High
011110
AF2
AF21
Short
010010
AF22
Medium
010100
AF23
High
010110
AF1
AF11
Short
001010
AF12
Medium
001100
AF13
High
001110
Table 2.8 AF DSCPs
The AF PHB ensures that packets are forwarded with a high probability of delivery to the destination within the bounds of the rate agreed upon in an SLA. If AF traffic at an ingress port exceeds the pre-priority rate, which is considered non-compliant or “out of profile”, the excess packets will not be delivered to the destination with the same probability as the packets belonging to the defined traffic or “in profile” packets. When there is network congestion, the out of profile packets are dropped before the in profile packets are dropped.
When service levels are defined using AF classes, different quantity and quality between AF classes can be realized by allocating different amounts of bandwidth and buffer space to the four AF classes. Unlike
EF, most AF traffic is non-real-time traffic using TCP, and the RED queue management strategy is an AQM (Adaptive Queue Management) strategy suitable for use in AF PHBs. The four AF PHB layers can be implemented as four separate queues. The output port bandwidth is divided into four AF queues. For each AF queue, packets are marked with three “colors” corresponding to three separate priority levels.
In addition to the 32 DSCP 1 groups defined in Table 2.8, 21 DSCPs have been standardized as follows: one for PHB EF, 12 for PHB AF, and 8 for CSCP. There are 11 DSCP 1 groups still available for other standards.
2.2.5.Example of Differentiated Services
We will look at an example of the Differentiated Service model and mechanism of operation. The architecture of Differentiated Service consists of two basic sets of functions:
Edge functions: include packet classification and traffic conditioning. At the inbound edge of the network, incoming packets are marked. In particular, the DS field in the packet header is set to a certain value. For example, in Figure 2.12, packets sent from H1 to H3 are marked at R1, while packets from H2 to H4 are marked at R2. The labels on the received packets identify the service class to which they belong. Different traffic classes receive different services in the core network. The RFC definition uses the term behavior aggregate rather than the term traffic class. After being marked, a packet can be forwarded immediately into the network, delayed for a period of time before being forwarded, or dropped. We will see that there are many factors that affect how a packet is marked, and whether it is forwarded immediately, delayed, or dropped.
Figure 2.12 DiffServ Example
Core functionality: When a DS-marked packet arrives at a Diffservcapable router, the packet is forwarded to the next router based on
Per-hop behavior is associated with packet classes. Per-hop behavior affects router buffers and the bandwidth shared between competing classes. An important principle of the Differentiated Service architecture is that a router's per-hop behavior is based only on the packet's marking or the class to which it belongs. Therefore, if packets sent from H1 to H3 as shown in the figure receive the same marking as packets from H2 to H4, then the network routers treat the packets exactly the same, regardless of whether the packet originated from H1 or H2. For example, R3 does not distinguish between packets from h1 and H2 when forwarding packets to R4. Therefore, the Differentiated Service architecture avoids the need to maintain router state about separate source-destination pairs, which is important for network scalability.
Chapter Conclusion
Chapter 2 has presented and clarified two main models of deploying and installing quality of service in IP networks. While the traditional best-effort model has many disadvantages, later models such as IntServ and DiffServ have partly solved the problems that best-effort could not solve. IntServ follows the direction of ensuring quality of service for each separate flow, it is built similar to the circuit switching model with the use of the RSVP resource reservation protocol. IntSer is suitable for services that require fixed bandwidth that is not shared such as VoIP services, multicast TV services. However, IntSer has disadvantages such as using a lot of network resources, low scalability and lack of flexibility. DiffServ was born with the idea of solving the disadvantages of the IntServ model.
DiffServ follows the direction of ensuring quality based on the principle of hop-by-hop behavior based on the priority of marked packets. The policy for different types of traffic is decided by the administrator and can be changed according to reality, so it is very flexible. DiffServ makes better use of network resources, avoiding idle bandwidth and processing capacity on routers. In addition, the DifServ model can be deployed on many independent domains, so the ability to expand the network becomes easy.
Chapter 3: METHODS TO ENSURE QoS FOR MULTIMEDIA COMMUNICATIONS
In packet-switched networks, different packet flows often have to share the transmission medium all the way to the destination station. To ensure the fair and efficient allocation of bandwidth to flows, appropriate serving mechanisms are required at network nodes, especially at gateways or routers, where many different data flows often pass through. The scheduler is responsible for serving packets of the selected flow and deciding which packet will be served next. Here, a flow is understood as a set of packets belonging to the same priority class, or originating from the same source, or having the same source and destination addresses, etc.
In normal state when there is no congestion, packets will be sent as soon as they are delivered. In case of congestion, if QoS assurance methods are not applied, prolonged congestion can cause packet drops, affecting service quality. In some cases, congestion is prolonged and widespread in the network, which can easily lead to the network being "frozen", or many packets being dropped, seriously affecting service quality.
Therefore, in this chapter, in sections 3.2 and 3.3, we introduce some typical network traffic load monitoring techniques to predict and prevent congestion before it occurs through the measure of dropping (removing) packets early when there are signs of impending congestion.
3.1. DropTail method
DropTail is a simple, traditional queue management method based on FIFO mechanism. All incoming packets are placed in the queue, when the queue is full, the later packets are dropped.
Due to its simplicity and ease of implementation, DropTail has been used for many years on Internet router systems. However, this algorithm has the following disadvantages:
− Cannot avoid the phenomenon of “Lock out”: Occurs when 1 or several traffic streams monopolize the queue, making packets of other connections unable to pass through the router. This phenomenon greatly affects reliable transmission protocols such as TCP. According to the anti-congestion algorithm, when locked out, the TCP connection stream will reduce the window size and reduce the packet transmission speed exponentially.
− Can cause Global Synchronization: This is the result of a severe “Lock out” phenomenon. Some neighboring routers have their queues monopolized by a number of connections, causing a series of other TCP connections to be unable to pass through and simultaneously reducing the transmission speed. After those monopolized connections are temporarily suspended,
Once the queue is cleared, it takes a considerable amount of time for TCP connections to return to their original speed.
− Full Queue phenomenon: Data transmitted on the Internet often has an explosion, packets arriving at the router are often in clusters rather than in turn. Therefore, the operating mechanism of DropTail makes the queue easily full for a long period of time, leading to the average delay time of large packets. To avoid this phenomenon, with DropTail, the only way is to increase the router's buffer, this method is very expensive and ineffective.
− No QoS guarantee: With the DropTail mechanism, there is no way to prioritize important packets to be transmitted through the router earlier when all are in the queue. Meanwhile, with multimedia communication, ensuring connection and stable speed is extremely important and the DropTail algorithm cannot satisfy.
The problem of choosing the buffer size of the routers in the network is to “absorb” short bursts of traffic without causing too much queuing delay. This is necessary in bursty data transmission. The queue size determines the size of the packet bursts (traffic spikes) that we want to be able to transmit without being dropped at the routers.
In IP-based application networks, packet dropping is an important mechanism for indirectly reporting congestion to end stations. A solution that prevents router queues from filling up while reducing the packet drop rate is called dynamic queue management.
3.2. Random elimination method – RED
3.2.1 Overview
RED (Random Early Detection of congestion; Random Early Drop) is one of the first AQM algorithms proposed in 1993 by Sally Floyd and Van Jacobson, two scientists at the Lawrence Berkeley Laboratory of the University of California, USA. Due to its outstanding advantages compared to previous queue management algorithms, RED has been widely installed and deployed on the Internet.
The most fundamental point of their work is that the most effective place to detect congestion and react to it is at the gateway or router.
Source entities (senders) can also do this by estimating end-to-end delay, throughput variability, or the rate of packet retransmissions due to drop. However, the sender and receiver view of a particular connection cannot tell which gateways on the network are congested, and cannot distinguish between propagation delay and queuing delay. Only the gateway has a true view of the state of the queue, the link share of the connections passing through it at any given time, and the quality of service requirements of the
traffic flows. The RED gateway monitors the average queue length, which detects early signs of impending congestion (average queue length exceeding a predetermined threshold) and reacts appropriately in one of two ways:
− Drop incoming packets with a certain probability, to indirectly inform the source of congestion, the source needs to reduce the transmission rate to keep the queue from filling up, maintaining the ability to absorb incoming traffic spikes.
− Mark “congestion” with a certain probability in the ECN field in the header of TCP packets to notify the source (the receiving entity will copy this bit into the acknowledgement packet).
Figure 3. 1 RED algorithm
The main goal of RED is to avoid congestion by keeping the average queue size within a sufficiently small and stable region, which also means keeping the queuing delay sufficiently small and stable. Achieving this goal also helps: avoid global synchronization, not resist bursty traffic flows (i.e. flows with low average throughput but high volatility), and maintain an upper bound on the average queue size even in the absence of cooperation from transport layer protocols.
To achieve the above goals, RED gateways must do the following:
− The first is to detect congestion early and react appropriately to keep the average queue size small enough to keep the network operating in the low latency, high throughput region, while still allowing the queue size to fluctuate within a certain range to absorb short-term fluctuations. As discussed above, the gateway is the most appropriate place to detect congestion and is also the most appropriate place to decide which specific connection to report congestion to.
− The second thing is to notify the source of congestion. This is done by marking and notifying the source to reduce traffic. Normally the RED gateway will randomly drop packets. However, if congestion
If congestion is detected before the queue is full, it should be combined with packet marking to signal congestion. The RED gateway has two options: drop or mark; where marking is done by marking the ECN field of the packet with a certain probability, to signal the source to reduce the traffic entering the network.
− An important goal that RED gateways need to achieve is to avoid global synchronization and not to resist traffic flows that have a sudden characteristic. Global synchronization occurs when all connections simultaneously reduce their transmission window size, leading to a severe drop in throughput at the same time. On the other hand, Drop Tail or Random Drop strategies are very sensitive to sudden flows; that is, the gateway queue will often overflow when packets from these flows arrive. To avoid these two phenomena, gateways can use special algorithms to detect congestion and decide which connections will be notified of congestion at the gateway. The RED gateway randomly selects incoming packets to mark; with this method, the probability of marking a packet from a particular connection is proportional to the connection's shared bandwidth at the gateway.
− Another goal is to control the average queue size even without cooperation from the source entities. This can be done by dropping packets when the average size exceeds an upper threshold (instead of marking it). This approach is necessary in cases where most connections have transmission times that are less than the round-trip time, or where the source entities are not able to reduce traffic in response to marking or dropping packets (such as UDP flows).
3.2.2 Algorithm
This section describes the algorithm for RED gateways. RED gateways calculate the average queue size using a low-pass filter. This average queue size is compared with two thresholds: minth and maxth. When the average queue size is less than the lower threshold, no incoming packets are marked or dropped; when the average queue size is greater than the upper threshold, all incoming packets are dropped. When the average queue size is between minth and maxth, each incoming packet is marked or dropped with a probability pa, where pa is a function of the average queue size avg; the probability of marking or dropping a packet for a particular connection is proportional to the bandwidth share of that connection at the gateway. The general algorithm for a RED gateway is described as follows: [5]
For each packet arrival
Caculate the average queue size avg If minth ≤ avg < maxth
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The Preposition “Auf” Viewed From a Cognitive Perspective Compared with Vietnamese -
Example Illustrating a Summary of an English Text -
The relationship between travel motivation, destination image and destination choice - A case study of Binh Dinh province tourism destination - 1
Among the four research perspectives mentioned above, Brown and Levinson's theory is considered the most influential. After Brown and Levinson's book was republished, many researchers from all over the world voiced their support. They conducted research in an applied direction based on Brown and Levinson's theoretical model. However, many researchers disagreed with Brown and Levinson's point of view. These representatives came from Eastern cultures such as China, Japan, and Korea. Typical authors are: Ide (1989), Hill Etal (1986), Matsumoto (1988), Gu (1990)... According to them, in cultures that emphasize community like the East, politeness is first and foremost a standard behavior, in accordance with the rules and order of hierarchy in society.

1.1.1.2. New approaches to politeness
A number of scholars have proposed new approaches to politeness. This trend has been most evident in the last ten years or so. The new research directions can be summarized as follows:
The post- modern approach with typical authors: Eelen (2001), Mills. S. (2003), Watts (2003), Locher and Watts (2005), Locher and Bousfield (2008), Bousfield and Culpepper (2008). The difference between this approach and the traditional approach is shown in the following aspects.
First, it is a rejection of Grice's traditional framework, which emphasizes the mediation of harmonious relationships and the speaker's intentions, and what is perceived by the hearer. They argue that Grice's framework is inappropriate to account for conflictual/antagonistic exchanges, such as are common in real life ('it is inappropriate to account for conflictual/antagonistic exchanges, such as are common in real life' [Dt 120; 241]. The postmodern tendency is also directed towards the hearer, placing politeness in the listener's judgment rather than the speaker's intentions. The tendency
This approach also rejects speech act theory (Mills 2003). The authors argue that politeness cannot focus on individual utterances but must be oriented towards longer discourses. In this respect, the postmodern tendency emphasizes the need to study conversation in a process-oriented view. [122; p. 38].
The relational approach with typical authors: Watts (2003) Locher (2004, 2006), Locher and Watts (2005), Spencer-Oatey (2005, 2007). Locher and Watts wrote: “Relational regulation activity is understood as people's investment in regulating their relationships in communication” [118; p. 78]. Although they may have different names such as: “relational work” (Locher and Watts 2005), “relational practice” (Holmes and Schnurr 2005) or “rapport management” (Spencer-Oatey 2000), they all have one thing in common: emphasizing interpersonal relationships rather than emphasizing the individual expression of politeness like traditional politeness models. [Dt 97; p. 22]. Similarly, impoliteness is also viewed by these linguists within the context of relationship regulation, avoiding presenting it as simply a dichotomy of politeness.
The frame -based view is a typical author, Marina Terkourafi (2001, 2002, 2003, 2005a, 2005b). This approach is a complement to the traditional and postmodern approaches. If the two approaches above take theory as the basis for research (theory-driven), the frame-based approach takes data as the basis for research (data-driven). Taking data from a large set of spontaneous conversations of people from Cyprus (Greece), Terkourafi strongly opposed Brown and Levinson's definition of politeness: "Politeness is a matter not of rational calculation, but a habit" (130; p. 250)
There is also the interactive approach with typical authors: Arundale (1999, 2006), Haugh (2007) and the genre approach with typical authors Garcés – Conejos Blitvich (2010a). [Dt 97; p. 26]
In general, the above approaches are quite new and have not had time to be thoroughly studied theoretically in all aspects and have not been tested practically like traditional research methods, however the initial results are very remarkable. It suggests many different approaches in this complex and also very attractive field.
The above is a brief overview of two research trends on politeness in the world. The traditional research direction is based on Grice's collaborative perspective, characterized by emphasizing the speaker and focusing on the analysis of individual utterances. The postmodern research direction focuses on the listener's evaluation and considers politeness in a more complete discourse. Each research direction has its own strengths, creating a colorful picture in the approach to the phenomenon of politeness.
The thesis chooses the politeness theory of Brown and Levinson as the theoretical basis for research. Although there are some conflicting opinions surrounding the politeness theory of these two authors, this is still considered the most influential theory ever.
1.1.2. The situation of research on politeness in Vietnam
The concept of politeness has been mentioned for a long time in the world, but it was not until the 60s and 70s of the 20th century that it was truly raised to a theory and became the main research object of pragmatics. In Vietnam, at first, people only saw its "shadow" through some related concepts such as: social role , face ... The first person to mention politeness was the author
Nguyen Dinh Hoa (1956) in Linguistic and non-linguistic models of polite behavior . Although he did not systematically analyze the theory of politeness, he introduced its premise concept - face. He analyzed the correlation between face and social behavior, in which face is understood as "pride in the values that one has" [Dt 14; p. 46]. In Social role and language behavior in communication [85], author Nhu Y studied the concept of social role as a factor governing the principle of politeness in communication: In reality, people are always in a position of diverse communication relationships with many different classes and types of people in terms of social status, age, gender, occupation, education, and social prestige. That social role governs the way individuals use language in communication, the standards of politeness, natural informality or gentleness, modesty... The theory of politeness is officially mentioned in On the study of politeness in communication [17] by author Nguyen Van Do. He summarized four trends of politeness research in the world: social standards perspective, conversational rules perspective, face perspective and conversational collaboration perspective and pointed out an open direction in politeness research: "Politeness has opened a new direction in language in general and in teaching and learning foreign languages in particular. If we can study and exploit it, we will certainly achieve new achievements in language research and teaching Vietnamese" [17; p. 57].
Along with the development of pragmatics, in Vietnam in recent years, many research works on politeness have appeared, both from a theoretical and practical perspective. Textbooks on pragmatics such as: General linguistics [6] by Do Huu Chau, Pragmatics [7] by Nguyen Duc Dan, Vietnamese language pragmatics
[21] by Nguyen Thien Giap; the sociolinguistics research work of author Nguyen Van Khang [39],… all mention the principle of politeness as a rule governing interpersonal relationships in conversation. The common point of these books is that they introduce the most basic concepts of
politeness: face, politeness, major schools of study, negative politeness, positive politeness, face-threatening and honoring behaviors,....
Politeness is also studied in connection with daily communication activities. The first research work on a general scale on the issue of politeness in Vietnamese is by author Vu Thi Thanh Huong. In her doctoral thesis Politeness in modern Vietnamese: A sociolinguistic study of a Hanoi speech community [132], using the method of investigation and testing with a number of subjects living in Hanoi, the author has proposed four main characteristics in the concept of Vietnamese people: politeness, propriety, tact, and tact. According to the author, these four characteristics have a relationship that includes each other but is not identical, both including and different.
Politeness is studied in relation to the OT. Almost every research work on the OT "touches" the issue of politeness. That is the thesis on politeness through giving and receiving behavior by author Chu Thi Bich [5]; the research work on complimenting behavior by Nguyen Quang [56], the research on criticizing behavior by Hoang Thi Hai Yen [87]; the research on greeting, thanking, apologizing utterances by Pham Thi Thanh [71],... Politeness is studied in a number of common communication rituals such as: inviting, thanking, congratulating, praising, apologizing, criticizing, rejecting, refusing,... as in the book Politeness in Vietnamese communication [68] by Ta Thi Thanh Tam. In the OT, the act of asking in relation to politeness is studied the most. (The research works of authors Nguyen Duc Hoat [28], Vu Thi Thanh Huong [32] can be mentioned. According to the authors, the reason why requests are of most interest is because "it is a type of behavior with a high level of face-threatening, so when performing it, politeness becomes the main concern" [32; p. 35]. In the above works, the authors not only raise the relationship between the OT and the
politeness but also analyzes the linguistic elements expressing politeness such as: address words, modal particles, barrier expressions, predicates... From there, each author contributes to clarifying the concept of politeness of Vietnamese people in communication practice.
Regarding the means of words expressing politeness, almost every research work on politeness has mentioned it, although it is still sporadic and scattered. If we only count the research works specifically on the means of words expressing politeness, we must mention the works of authors Vu Tien Dung [10], [12], Nguyen Thi Luong [43], Vu Thi Nga [46], ... Here, the words have been meticulously analyzed to find out their roles, functions as well as their different levels in expressing politeness.
In addition, politeness is also viewed in relation to a number of other factors such as gender, age, occupation, etc. Most notably, the relationship between politeness and gender. This issue is mentioned in the studies of authors Vu Tien Dung [12], Vu Thi Thanh Huong [33], etc. Politeness is compared to its close concept - politeness in the thesis of Phan Thi Phuong Dung [8].
Politeness is also studied from the perspective of cross-cultural pragmatics. The concept of politeness is not only viewed within a cultural community but is compared and contrasted in different cultural and linguistic communities. For example, comparing communication protocols between the English and Vietnamese speaking communities, Japanese and Vietnamese in the light of politeness theory. Research works and articles that must be mentioned are: Politeness in critical speech acts of Vietnamese and British people [23] by Le Thi Thuy Ha, Politeness and means of expressing politeness in Vietnamese and Japanese requests [55] by Tran Lan Phuong, Communication and cross-cultural communication [57], Some issues of intracultural and cross-cultural communication [58] by
Author Nguyen Quang; Comparing and contrasting politeness in communication in Thai and Vietnamese [62] by author Siriwong Hongsawan...; Linguistic means of expressing politeness in Vietnamese and Japanese [75] by Hoang Anh Thi (1998),... This research direction promises many interesting things, with practical significance in foreign language teaching and learning activities.
The issue of politeness in journalistic interviews has been addressed in several recent theses such as: Questioning in television interview language
[80] by Tran Phuc Trung, Conversational language in the interview genre (on current Vietnamese printed newspaper materials) [31] by author Pham Thi Mai Huong. When analyzing the act of asking in a television interview, author Tran Phuc Trung mentioned politeness as a characteristic in the behavioral culture of interview communication. In which, the author raised a number of strategies to increase politeness for the act of asking in a television interview, including: Politeness in the use of modal words like "ạ", "ạ", "vữu "; Politeness in the way of addressing; Politeness in apologies related to asking. The thesis only mentioned a few manifestations of politeness in a television interview. Politeness has not been systematically studied as a main research object on interview communication conversation materials.
In general, politeness has been studied from many angles, in relation to many factors, with many elaborate research works, with high generality. That is the basis, the advantage for the thesis in inheriting, and at the same time, it is also a big challenge, how to have discovery, innovation compared to previous works. Looking at the general research works on politeness, it can be seen that politeness has been exploited more in the positive aspect - compliance (politeness) but less attention is paid to the negative aspect - violation (impoliteness). Besides, politeness is mainly studied on the text of daily conversation. Selecting research text


![Qos Assurance Methods for Multimedia Communications
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low. The EF PHB requires a sufficiently large number of output ports to provide low delay, low loss, and low jitter.
EF PHBs can be implemented if the output ports bandwidth is sufficiently large, combined with small buffer sizes and other network resources dedicated to EF packets, to allow the routers service rate for EF packets on an output port to exceed the arrival rate λ of packets at that port.
This means that packets with PHB EF are considered with a pre-allocated amount of output bandwidth and a priority that ensures minimum loss, minimum delay and minimum jitter before being put into operation.
PHB EF is suitable for channel simulation, leased line simulation, and real-time services such as voice, video without compromising on high loss, delay and jitter values.
Figure 2.10 Example of EF installation
Figure 2.10 shows an example of an EF PHB implementation. This is a simple priority queue scheduling technique. At the edges of the DS domain, EF packet traffic is prioritized according to the values agreed upon by the SLA. The EF queue in the figure needs to output packets at a rate higher than the packet arrival rate λ. To provide an EF PHB over an end-to-end DS domain, bandwidth at the output ports of the core routers needs to be allocated in advance to ensure the requirement μ > λ. This can be done by a pre-configured provisioning process. In the figure, EF packets are placed in the priority queue (the upper queue). With such a length, the queue can operate with μ > λ.
Since EF was primarily used for real-time services such as voice and video, and since real-time services use UDP instead of TCP, RED is generally
not suitable for EF queues because applications using UDP will not respond to random packet drop and RED will strip unnecessary packets.
2.2.4.2 Assured Forwarding (AF) PHB
PHB AF is defined by RFC 2597. The purpose of PHB AF is to deliver packets reliably and therefore delay and jitter are considered less important than packet loss. PHB AF is suitable for non-real-time services such as applications using TCP. PHB AF first defines four classes: AF1, AF2, AF3, AF4. For each of these AF classes, packets are then classified into three subclasses with three distinct priority levels.
Table 2.8 shows the four AF classes and 12 AF subclasses and the DSCP values for the 12 AF subclasses defined by RFC 2597. RFC 2597 also allows for more than three separate priority levels to be added for internal use. However, these separate priority levels will only have internal significance.
PHB Class
PHB Subclass
Package type
DSCP
AF4
AF41
Short
100010
AF42
Medium
100100
AF43
High
100110
AF3
AF31
Short
011010
AF32
Medium
011100
AF33
High
011110
AF2
AF21
Short
010010
AF22
Medium
010100
AF23
High
010110
AF1
AF11
Short
001010
AF12
Medium
001100
AF13
High
001110
Table 2.8 AF DSCPs
The AF PHB ensures that packets are forwarded with a high probability of delivery to the destination within the bounds of the rate agreed upon in an SLA. If AF traffic at an ingress port exceeds the pre-priority rate, which is considered non-compliant or “out of profile”, the excess packets will not be delivered to the destination with the same probability as the packets belonging to the defined traffic or “in profile” packets. When there is network congestion, the out of profile packets are dropped before the in profile packets are dropped.
When service levels are defined using AF classes, different quantity and quality between AF classes can be realized by allocating different amounts of bandwidth and buffer space to the four AF classes. Unlike
EF, most AF traffic is non-real-time traffic using TCP, and the RED queue management strategy is an AQM (Adaptive Queue Management) strategy suitable for use in AF PHBs. The four AF PHB layers can be implemented as four separate queues. The output port bandwidth is divided into four AF queues. For each AF queue, packets are marked with three “colors” corresponding to three separate priority levels.
In addition to the 32 DSCP 1 groups defined in Table 2.8, 21 DSCPs have been standardized as follows: one for PHB EF, 12 for PHB AF, and 8 for CSCP. There are 11 DSCP 1 groups still available for other standards.
2.2.5.Example of Differentiated Services
We will look at an example of the Differentiated Service model and mechanism of operation. The architecture of Differentiated Service consists of two basic sets of functions:
Edge functions: include packet classification and traffic conditioning. At the inbound edge of the network, incoming packets are marked. In particular, the DS field in the packet header is set to a certain value. For example, in Figure 2.12, packets sent from H1 to H3 are marked at R1, while packets from H2 to H4 are marked at R2. The labels on the received packets identify the service class to which they belong. Different traffic classes receive different services in the core network. The RFC definition uses the term behavior aggregate rather than the term traffic class. After being marked, a packet can be forwarded immediately into the network, delayed for a period of time before being forwarded, or dropped. We will see that there are many factors that affect how a packet is marked, and whether it is forwarded immediately, delayed, or dropped.
Figure 2.12 DiffServ Example
Core functionality: When a DS-marked packet arrives at a Diffservcapable router, the packet is forwarded to the next router based on
Per-hop behavior is associated with packet classes. Per-hop behavior affects router buffers and the bandwidth shared between competing classes. An important principle of the Differentiated Service architecture is that a routers per-hop behavior is based only on the packets marking or the class to which it belongs. Therefore, if packets sent from H1 to H3 as shown in the figure receive the same marking as packets from H2 to H4, then the network routers treat the packets exactly the same, regardless of whether the packet originated from H1 or H2. For example, R3 does not distinguish between packets from h1 and H2 when forwarding packets to R4. Therefore, the Differentiated Service architecture avoids the need to maintain router state about separate source-destination pairs, which is important for network scalability.
Chapter Conclusion
Chapter 2 has presented and clarified two main models of deploying and installing quality of service in IP networks. While the traditional best-effort model has many disadvantages, later models such as IntServ and DiffServ have partly solved the problems that best-effort could not solve. IntServ follows the direction of ensuring quality of service for each separate flow, it is built similar to the circuit switching model with the use of the RSVP resource reservation protocol. IntSer is suitable for services that require fixed bandwidth that is not shared such as VoIP services, multicast TV services. However, IntSer has disadvantages such as using a lot of network resources, low scalability and lack of flexibility. DiffServ was born with the idea of solving the disadvantages of the IntServ model.
DiffServ follows the direction of ensuring quality based on the principle of hop-by-hop behavior based on the priority of marked packets. The policy for different types of traffic is decided by the administrator and can be changed according to reality, so it is very flexible. DiffServ makes better use of network resources, avoiding idle bandwidth and processing capacity on routers. In addition, the DifServ model can be deployed on many independent domains, so the ability to expand the network becomes easy.
Chapter 3: METHODS TO ENSURE QoS FOR MULTIMEDIA COMMUNICATIONS
In packet-switched networks, different packet flows often have to share the transmission medium all the way to the destination station. To ensure the fair and efficient allocation of bandwidth to flows, appropriate serving mechanisms are required at network nodes, especially at gateways or routers, where many different data flows often pass through. The scheduler is responsible for serving packets of the selected flow and deciding which packet will be served next. Here, a flow is understood as a set of packets belonging to the same priority class, or originating from the same source, or having the same source and destination addresses, etc.
In normal state when there is no congestion, packets will be sent as soon as they are delivered. In case of congestion, if QoS assurance methods are not applied, prolonged congestion can cause packet drops, affecting service quality. In some cases, congestion is prolonged and widespread in the network, which can easily lead to the network being frozen, or many packets being dropped, seriously affecting service quality.
Therefore, in this chapter, in sections 3.2 and 3.3, we introduce some typical network traffic load monitoring techniques to predict and prevent congestion before it occurs through the measure of dropping (removing) packets early when there are signs of impending congestion.
3.1. DropTail method
DropTail is a simple, traditional queue management method based on FIFO mechanism. All incoming packets are placed in the queue, when the queue is full, the later packets are dropped.
Due to its simplicity and ease of implementation, DropTail has been used for many years on Internet router systems. However, this algorithm has the following disadvantages:
− Cannot avoid the phenomenon of “Lock out”: Occurs when 1 or several traffic streams monopolize the queue, making packets of other connections unable to pass through the router. This phenomenon greatly affects reliable transmission protocols such as TCP. According to the anti-congestion algorithm, when locked out, the TCP connection stream will reduce the window size and reduce the packet transmission speed exponentially.
− Can cause Global Synchronization: This is the result of a severe “Lock out” phenomenon. Some neighboring routers have their queues monopolized by a number of connections, causing a series of other TCP connections to be unable to pass through and simultaneously reducing the transmission speed. After those monopolized connections are temporarily suspended,
Once the queue is cleared, it takes a considerable amount of time for TCP connections to return to their original speed.
− Full Queue phenomenon: Data transmitted on the Internet often has an explosion, packets arriving at the router are often in clusters rather than in turn. Therefore, the operating mechanism of DropTail makes the queue easily full for a long period of time, leading to the average delay time of large packets. To avoid this phenomenon, with DropTail, the only way is to increase the routers buffer, this method is very expensive and ineffective.
− No QoS guarantee: With the DropTail mechanism, there is no way to prioritize important packets to be transmitted through the router earlier when all are in the queue. Meanwhile, with multimedia communication, ensuring connection and stable speed is extremely important and the DropTail algorithm cannot satisfy.
The problem of choosing the buffer size of the routers in the network is to “absorb” short bursts of traffic without causing too much queuing delay. This is necessary in bursty data transmission. The queue size determines the size of the packet bursts (traffic spikes) that we want to be able to transmit without being dropped at the routers.
In IP-based application networks, packet dropping is an important mechanism for indirectly reporting congestion to end stations. A solution that prevents router queues from filling up while reducing the packet drop rate is called dynamic queue management.
3.2. Random elimination method – RED
3.2.1 Overview
RED (Random Early Detection of congestion; Random Early Drop) is one of the first AQM algorithms proposed in 1993 by Sally Floyd and Van Jacobson, two scientists at the Lawrence Berkeley Laboratory of the University of California, USA. Due to its outstanding advantages compared to previous queue management algorithms, RED has been widely installed and deployed on the Internet.
The most fundamental point of their work is that the most effective place to detect congestion and react to it is at the gateway or router.
Source entities (senders) can also do this by estimating end-to-end delay, throughput variability, or the rate of packet retransmissions due to drop. However, the sender and receiver view of a particular connection cannot tell which gateways on the network are congested, and cannot distinguish between propagation delay and queuing delay. Only the gateway has a true view of the state of the queue, the link share of the connections passing through it at any given time, and the quality of service requirements of the
traffic flows. The RED gateway monitors the average queue length, which detects early signs of impending congestion (average queue length exceeding a predetermined threshold) and reacts appropriately in one of two ways:
− Drop incoming packets with a certain probability, to indirectly inform the source of congestion, the source needs to reduce the transmission rate to keep the queue from filling up, maintaining the ability to absorb incoming traffic spikes.
− Mark “congestion” with a certain probability in the ECN field in the header of TCP packets to notify the source (the receiving entity will copy this bit into the acknowledgement packet).
Figure 3. 1 RED algorithm
The main goal of RED is to avoid congestion by keeping the average queue size within a sufficiently small and stable region, which also means keeping the queuing delay sufficiently small and stable. Achieving this goal also helps: avoid global synchronization, not resist bursty traffic flows (i.e. flows with low average throughput but high volatility), and maintain an upper bound on the average queue size even in the absence of cooperation from transport layer protocols.
To achieve the above goals, RED gateways must do the following:
− The first is to detect congestion early and react appropriately to keep the average queue size small enough to keep the network operating in the low latency, high throughput region, while still allowing the queue size to fluctuate within a certain range to absorb short-term fluctuations. As discussed above, the gateway is the most appropriate place to detect congestion and is also the most appropriate place to decide which specific connection to report congestion to.
− The second thing is to notify the source of congestion. This is done by marking and notifying the source to reduce traffic. Normally the RED gateway will randomly drop packets. However, if congestion
If congestion is detected before the queue is full, it should be combined with packet marking to signal congestion. The RED gateway has two options: drop or mark; where marking is done by marking the ECN field of the packet with a certain probability, to signal the source to reduce the traffic entering the network.
− An important goal that RED gateways need to achieve is to avoid global synchronization and not to resist traffic flows that have a sudden characteristic. Global synchronization occurs when all connections simultaneously reduce their transmission window size, leading to a severe drop in throughput at the same time. On the other hand, Drop Tail or Random Drop strategies are very sensitive to sudden flows; that is, the gateway queue will often overflow when packets from these flows arrive. To avoid these two phenomena, gateways can use special algorithms to detect congestion and decide which connections will be notified of congestion at the gateway. The RED gateway randomly selects incoming packets to mark; with this method, the probability of marking a packet from a particular connection is proportional to the connections shared bandwidth at the gateway.
− Another goal is to control the average queue size even without cooperation from the source entities. This can be done by dropping packets when the average size exceeds an upper threshold (instead of marking it). This approach is necessary in cases where most connections have transmission times that are less than the round-trip time, or where the source entities are not able to reduce traffic in response to marking or dropping packets (such as UDP flows).
3.2.2 Algorithm
This section describes the algorithm for RED gateways. RED gateways calculate the average queue size using a low-pass filter. This average queue size is compared with two thresholds: minth and maxth. When the average queue size is less than the lower threshold, no incoming packets are marked or dropped; when the average queue size is greater than the upper threshold, all incoming packets are dropped. When the average queue size is between minth and maxth, each incoming packet is marked or dropped with a probability pa, where pa is a function of the average queue size avg; the probability of marking or dropping a packet for a particular connection is proportional to the bandwidth share of that connection at the gateway. The general algorithm for a RED gateway is described as follows: [5]
For each packet arrival
Caculate the average queue size avg If minth ≤ avg < maxth
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