different ways to highlight important and noteworthy information segments in the text, thereby guiding listeners to correctly perceive the role of that important information, focus their attention on that information appropriately and process it accurately .
With this focus of information, focus functions like a spotlight. It selects a particular piece of information and declares, “This piece of information is of particular importance.” In other words, the speaker announces to the listener, “This is important, please listen!”
In short, when discussing this linguistic phenomenon, we want to emphasize the effectiveness or effect of using focalization to help the speaker or writer focus on the information parts that are considered important in the message structure of the sentence and are expressed through many surface structure devices. These devices are very diverse in form. We will discuss it right after this.
1.2.3.2. Focus marking method
When talking about the phenomenon of focalization, we cannot help but mention the means of expression. There are three common types of language means used to express the focalization of information. These are phonetic-phonological means, lexical means and syntactic means.
(i) Phonetic and phonological means
Phonetically, focal information can be identified by means of stress and accent. It can be asserted that one of the ways to mark the focus of a sentence is to create a pitch. The speaker will "clarify the focus by pronouncing the words in the focus differently from the words that are not in the focus, including pronouncing them more tensely, more strongly, louder, or even unusually softly" [3,276]. And the usual position of the focus by accent is at the end of the sentence. For example:
[1:8] An broke the neighbor's glass .
But in our opinion, the focus of information is not necessarily on the last element of the sentence. For example:
[1:8a] An broke his neighbor's glass. ( An (not someone else) broke his neighbor's glass).
[1:8b] An broke his neighbor's glass. (The action that An did to his neighbor's glass was breaking it .)
[1:8c] An broke his neighbor's glass . (The glass that An broke belonged to his neighbor , not anyone else.)
[1:8d] An broke his neighbor's glass . (An broke the glass , not anything else.)
The phenomenon of focalization marking from the perspective of stress and intonation has been of interest to many linguists (see Quirk, R. [54]. Recently, author Chungmin Lee [53] has a research work on the contrasting ways of expressing focus and topic in English and Korean. However, they also emphasized that stress and intonation are not the only ways of expressing focus.
However, this issue requires sophisticated survey tools and research processes on phonetics and phonology, so in this thesis we only mention it without intending to delve too deeply into it.
(ii) Lexical means
The effect of focalizing a subject-predicate structure can be created when a word or phrase is added or reused in an utterance.
a. Add word
Words that create a focal point for information can be added before or after the word containing the highlighted information. For example:
[1:9] The picture of martyr Nguyen Van Be hangs right in the middle of the tent, next to a row of books.
(NTNT:455)
b. Repeat words
An expression containing a piece of information chosen as a focus can be repeated to demonstrate its salience. The repeated piece can be a word or phrase, and can be repeated in its entirety or with some variation. For example:
[1:10] The stars in the sky gave birth to her as a girl, a girl from Hang Dao Street ; a girl from Hang Dao Street with beauty ; a girl from Hang Dao Street with the beauty of a rich family ; a girl from Hang Dao Street with the beauty of a rich family in the prime of her youth, but she was not born into a family that was free to dress, to be worthy of all the things that she had that were better than others.
(NCH:147)
In Vietnamese, the lexical tools to create focal points are very diverse and rich (they can be emphatic auxiliary words such as chinh, ngay, ca, rieng ... modal particles such as co, day, ch... ; combinations of auxiliary words and particles such as chi... thoi, co... thi co ...) and always go together with structural tools to create a tight system. Therefore, when studying the tools to express focal points of the subject-predicate structure of a sentence, it is necessary to examine the syntactic means that we will mention below.
(iii) Syntactic means
Quirk suggests that "the best way to consider the location of the focus is to place it in relation to the syntactic structure of the sentence" [54,938]. He offers the following explanation for this problem: "The sentence (simple sentence) is a grammatical unit closely related to the intonation unit, or information unit" and therefore the information structure is conveyed through the syntactic structure of the sentence. In saying this, the author intends to emphasize that the syntactic structure of the sentence is the realization of the information structure and is the specific syntactic unit for examining the means of focus. Syntactic means are often expressed through syntactic transformations. Syntactic transformations are the operations of transforming from one structure to another without changing the relationship of the actual words participating in that transformation. The change in syntactic structure does not change the content of the sentence but changes its semantic-pragmatic value; thereby creating an information-emphasizing effect (information focalization). Here we see the asymmetrical nature of language, meaning that there is no one-to-one correspondence between the two sides of the language signal: the same form (the expression), can contain many contents (the expressed) and vice versa, the same content (the expressed) can be expressed in many forms (the expressed). Regarding the focalization phenomenon, we believe that the speaker does not create a new structure to express new information but only skillfully and creatively uses existing structures to effectively carry out the purpose of communication. Therefore, in each language in general and Vietnamese in particular, there always exist certain ways of expression. That is the reason why we want to study the means to mark the focalization of the subject-predicate structure of Vietnamese sentences.
In addition, to summarize and analyze the lexical and syntactic means used to create the effects of focal phenomena, we rely on some basic principles to establish some necessary and sufficient conditions for means of expressing focal information:
(a) Grice's cooperative principle [4]
The principle of conversational cooperation has a general form: Make your contribution (to the conversation) exactly as it is required at the stage (of the conversation) in which it appears, in accordance with the purpose or direction of the conversation in which you have agreed to participate (quoted from Do Huu Chau [4,229]). This principle has four categories: quantity, quality, relation, and manner in the spirit of the categories of the philosopher Kant. Each of these categories corresponds to a "sub-principle" that Grice called a maxim:
(i) Maxim of quantity:
- Make your contributions as informative as required (of the ongoing purpose of each part of the conversation)
- Don't make your contribution more informative than required.
(ii) Quality maxim
- Don't say things that you believe are not true.
- Don't say things you don't have solid evidence for.
(iii) Relationship maxim: Speak in the right place.
(iv) Manner maxim
- Avoid ambiguous language
- Avoid ambiguous language (which can have multiple meanings)
- Be brief (avoid wordy)
- Speak in order.
This principle applies to special sentence structures - as a syntactic means of expressing the focal phenomenon of the sentence. For example, instead of a structure with all its components and in a normal order such as Subject - Predicate - Complement, the reader may find a structure lacking the components: Adverb - Predicate - Subject
This inverted order, applying Grice's principle, is certainly for a certain purpose when the speaker deliberately introduces the adverbial part as the topic of his utterance.
Grice himself admitted that in addition to the above maxims, "others could be added" (4). He also asserted that among these maxims, some maxims should be respected more than others.
(b) Truth conditions
J.Lyons [28,157] when discussing propositions and propositional content stated the views of philosophers on four criteria of a normal proposition:
(i) either true or false;
(ii) may be known, believed or suspected;
(iii) may be confirmed, denied or questioned;
(iv) remains unchanged when translated from one language to another.
Information about propositional content is thus determined by the truth-condition of the utterance. Two different sentences expressing the same propositional content, i.e. having the same truth-condition, will either be true or false. In that case, we can say that the two sentences above carry the same semantic information content: propositional information.
On the basis of this concept, a medium is used to mark a point if and only if:
(i) the means does not change the truth conditions, or does not change the content of the proposition;
(ii) the device highlights notable pieces of information in the sentence;
(iii) the means must be marked.
In this thesis, we do not intend to study all the lexical and syntactic devices of Vietnamese, but only those lexical and syntactic devices that have the effect of creating focus and highlighting noteworthy information. Therefore, in the process of research, we found that there are terms that do not exist in Vietnamese, so we have to borrow them from English to translate into Vietnamese, such as the structure of inverting an element to the beginning of a sentence called preposing or inverting a certain element to the end of a sentence called postposing. This will be presented in detail in Chapter II.
1.2.4. Marking theory
1.2.4.1. Jakobson's theory of marking
Trubetzkoy was the first to discover the concept of marked/unmarked. This concept was later called "marked" and developed the theory of markedness in phonology by introducing three phonemic oppositions:
(i) Contrast with/without: that is, the contrast between two phonemes, all other distinguishing features are identical, they only differ in one distinguishing feature according to the two values of having or not having that distinguishing feature. He further explained that the marked component of a pair of contrasts is the phoneme characterized by the existence of the positive value ( of having ) of the distinguishing feature. The unmarked component is the component characterized by the absence ( of not having ) of that distinguishing feature. We can take the example of the pair /d/ and / t / then
/d/ is a marked component because it is voiced, and /t/ is unmarked because it lacks the voiced (voiceless) stroke.
(ii) Contrast of scale: that is, the contrast between a number of phonemes that are identical in all distinguishing features, differing only in different degrees of a certain distinguishing feature. For example, the contrast between the three phonemes /i/, /e/, /ε/ in Vietnamese. They are all front-row vowels, differing only in the degree of mouth opening: /i/ has the narrowest opening, then /e/ (written as ê). The widest is the vowel /ε/ (written as e).
(iii) Equivalent opposition: all other oppositions, not yes/no oppositions, not scale oppositions. In other words, each member of a group has a characteristic that other members of the same group do not have. For example, the two consonants /p/ and /t/ are both voiceless, stop consonants, but /p/ is a labial consonant, /t/ is a labial-dental consonant. The labial distinction is equivalent to the labial-dental distinction (this is not a yes/no opposition because the labial-dental distinction is not a labial distinction).
The phonemic structure of Trubetzkoy and of the Prague school was continued by R.Jakobson, an outstanding representative of this school. Around 1940, R.Jakobson further developed the theory of phonological differentiation. But the difference between Trubetzkoy and R.Jakobson was that Trubetzkoy, due to the limitations of the technical conditions of the time, only used articulatory characteristics, while R.Jakobson, thanks to quite sophisticated acoustic machines, used acoustic characteristics as phonological differentiation.
Studying a wide range of very different languages, Jakobson showed that only a few distinctive features, about 12, were sufficient to describe them.
On the other hand, Jakobson's theory of marking is a theory that deals with the relationship between marked and unmarked units in binary oppositions. He defines a marked unit as an utterance of a property A, while an unmarked unit can be divided into two parts: either a non-utterance of A or an utterance of A. A property is defined by Jakobson as a given semantic property that is relatively independent of extra-linguistic reality. He notes that markedness must be considered in relation to unmarked units.
For example, in English, count nouns have two forms: singular ( book ) and plural ( books ). The plural form is indicated by the explicit presence of an "-s" at the end of the word; the singular form by the absence of a similar inflection. Such a presence or absence of a formal marker corresponds to a semantic difference: the plural form ( books ) refers to more than one unit of books; but the singular form is not necessarily limited to one unit of meaning, as we find in many examples: bookshop, bookseller, book-shelf, bookstore , etc. Such cases are not necessarily plural, they can be singular or neuter. Since they are more likely to occur than the plural form, they are considered unmarked. All the unmarked elements in this example include all the cases of singular book and non-indicator book , and the opposite elements (the unmarked elements) are the cases of plural books. The pair of opposites above can be described as follows:
[1:12]book | Books | |
markup | unmarked | tick |
form | Absent suffix | suffix present |
meaning | non-plural | plural |
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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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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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Situation of Information Technology Application in Enterprises -
Perspectives on Improving the Quality of Law Application in Resolving Land Use Rights Disputes at the People's Court -
Concept of Law Application in First Instance Trials of Juvenile Criminals of the People's Court
1.2.4.2. Application of marking theory in functional grammar
Based on Jakobson's theory of markedness, Dik applied the terms marked and unmarked to language. According to Dik, cases with high frequency of occurrence are unmarked cases and cases with low frequency of occurrence are marked cases. He gives an example
about word order in English with the common structure: S + Vf + X (S= subject; Vf= main verb in agreement with the subject; X= other elements). Through a survey of 100 English sentences, he thought that inverted word order in English must certainly express a meaning other than the function of providing information.
Compared with the information structure of Vietnamese, the topic position is usually placed at the same position as the topic and subject, but when another element of the sentence is promoted to this position, we can mark and create an information focus in the utterance.
Dik also argued:
(i) A phenomenon that is considered marked in one (linguistic) environment may be unmarked in another (linguistic) environment;
(ii) When marked forms are used frequently, they gradually lose their markedness and a new marked form may emerge to replace them.
Dik's views on markedness have helped us gain a deeper insight into the phenomenon of focalization: the speaker/writer can use a certain word to replace or supplement another word to convey a message that needs to be emphasized or can use an unusual structure to express a special meaning, creating an informational focus in the sentence to attract the listener's attention (a focal structure) to replace an unmarked structure that has no informational focus or emphasis value.
We give an example that applies the theory of marking through the use of focalization method, which is a structure containing a determiner at the beginning of the sentence. In addition to adverbs used to separate sentences such as then/la/ma , Vietnamese sentences can have modal adverbs such as mai/moi to contribute to creating information focus. See example:
[1:11] It was not until he heard Hai Thep calling from in front of the cave that he stood up.
(AD 1:94 )
[1:12] Only the sound of water flowing impatiently from the stream echoed that statement.
(CL:18)