receive routing area updates ''). Since the routing frame is unchanged, it is not necessary to notify other network elements such as the GGSN or HLR.
Inter-SGSN routing area update: the new routing area is managed by another SGSN. The new SGSN notices that the mobile station has moved into its area and requests the old SGSN to send the subscriber's packet data protocol (PDP) frame. The new SGSN then informs the relevant GGSNs about the
the subscriber's new routing frame. In addition, the HLR and MSC/VLR (if any) will also be informed about the subscriber's new SGSN.
2.6.2.1. Updating routing areas within the SGSN
The following describes the routing area update procedure within the SGSN:
need)
t routing area
2. Security functions
3. Accept updates
4. Complete routing area update
MS
BSS
SGSN
1. Update request
routing area
Mandatory Optional or Conditional
Figure 2 – 13: Routing area update procedure within SGSN
1. The mobile station sends a routing area update request to the SGSN. The SGSN routing area update request contains information about the old routing area and the old P-TMSI number. The base station subsystem adds to the routing area update request the identification of the cell from which the BSS received the message and then forwards the routing area update request to the SGSN.
2. Security functions such as authentication and encryption can be performed.
3. The SGSN confirms the presence of the mobile station in the new routing area. If, due to regional subscriber restrictions, the mobile station is not allowed to join the network in the new routing area or there is a subscriber check error, the SGSN will reject the area update.
routing area for the mobile station. If all checks are successful, the SGSN updates the mobile station's mobility management frame. A new P-TMSI is assigned. A Routing Area Update message is sent back to the mobile station. The P-TMSI is included in the acceptance message.
4. After the P-TMSI number is assigned at the mobile station, the mobile station will confirm the new P-TMSI number with a routing area update completion message.
2.6.2.2. Updating routing areas between SGSNs
Figure 2 – 14 illustrates the routing area update procedure when a mobile station moves from one SGSN area to another SGSN area.
1. The mobile station sends a routing area update request to the new SGSN. The routing area update request contains information about the old routing area and the old P-TMSI number. The BSS adds to the routing area update request the identification of the cell from which the BSS received the message and then forwards the routing area update request to the SGSN.
2. The new SGSN sends an SGSN frame request to the old SGSN to retrieve the mobile management frames and packet data protocol frames for the mobile station. The old SGSN will check the old P-TMSI number and will respond with an error message if the P-TMSI number does not match the P-TMSI number stored in the old SGSN. At that time, the SGSN starts to perform the security functions. If the mobile station authentication function confirms the correctness of the mobile station. The new SGSN will send an SGSN frame request message to the old SGSN. If the P-TMSI number is correct or the SGSN has authenticated the mobile station, the old SGSN will reply to the new SGSN with an SGSN frame reply message. If the mobile station is not recognized by the old SGSN, the old SGSN will send a corresponding error message. The old SGSN stores the address of the new SGSN to allow the SGSN to forward data packets to the new SGSN. The old SGSN starts the counter and stops transmitting when the session ends on the downlink.
3. Security functions such as authentication and encryption are performed
4. The new SGSN sends an SGSN frame acknowledgement message to the old SGSN. The message will inform the old SGSN that the new SGSN is ready to receive data packets of the initiated packet data protocol frames. The old SGSN will mark in its frame that the MSC/VLR and the information in the GGSNs and HLRs are correct. The old SGSN will contact the MSC/VLRs, GGSNs and HLRs for updates when the mobile station initiates the routing area update procedure back to the old SGSN first.
when the current routing area update procedure is completed. If the security functions cannot determine the correctness of the mobile station, the routing area update procedure is rejected and the new SGSN sends a reject message to the old SGSN. The old SGSN proceeds as if the SGSN frame request message was never received.
MS BSS
SGSN
SGSN
GGSN
HLR
1. Request routing area update
2. SGSN frame request
2. SGSN frame request response
3. Security functions
SGSN
4. Frame confirmation
5. Packet transmission
6. Packet data protocol frame update request
6. Respond to packet data protocol frame update request
7. Update location
8. Delete location
8. Confirm deletion of location
9. Enter subscriber data
9. Confirm subscriber data entry
10. Confirm location update
11. Accept routing area updates
12. Complete routing area update
Mandatory procedures Optional or conditional procedures
Figure 2 – 14: Inter-SGSN routing area network entry procedure
5. The old SGSN will forward the data packets in the buffer to the new SGSN. The data packets will be received from the GGSN for forwarding to the new SGSN until the counter in step 2 stops counting. The packet data has been sent to the mobile station in acknowledgement mode as well as with the LLC frame number, which is sent at the end of the packet data frame. After
Once the counter in step 2 has stopped counting, no new packet data is sent to the new SGSN.
6. The new SGSN sends a Packet Data Protocol Frame Update Request message to the relevant GGSNs. The GGSNs will be updated with the Packet Data Protocol Frame fields and respond with a Packet Data Protocol Frame Update Request Reply message.
7. The new SGSN informs the HLR about the change of SGSN by sending a location update message to the HLR.
8. The HLR sends a Location Clear message to the old SGSN. If the counter in step 2 is not present, the old SGSN deletes the Mobility Management Frames and Packet Data Protocol Frames. If the counter in step 2 is present, the frames are deleted when the counter stops counting. The old SGSN then terminates the packet data transmission. This also allows the old SGSN to retain the Mobility Management Frames and Packet Data Protocol Frames in case the mobile station starts another SGSN Routing Area Update procedure before the current Routing Area Update procedure is completed. The old SGSN acknowledges this with a Location Clear Confirmation message.
9. The HLR sends the subscriber data to the new SGSN. The new SGSN checks the presence of the mobile station in the new routing area. If, due to area-based subscriber restrictions, the mobile station is not allowed to enter the routing area, the SGSN rejects the routing area update request with the cause of the error and sends a subscriber data import confirmation message to the HLR. If all checks are successful, the SGSN creates a mobility management frame for the mobile station and sends a subscriber data import confirmation message to the HLR.
10. The HLR confirms the location update by sending an update acknowledgement message.
locate to new SGSN.
11. The new SGSN will check the presence of the mobile station in the new routing area. If due to roaming subscriber restrictions, the mobile station is not allowed to join the network in the routing area or the subscriber check fails, the SGSN will send a routing area update reject message along with the corresponding error reason. If all checks are successful, the new SGSN will generate mobility management frames and packet data protocol frames to the mobile station. The new SGSN replies to the mobile station with a routing area update accept message.
12. The mobile station acknowledges the receipt of the new P-TMSI together with the Routing Area Update Complete message. If the Routing Area Update Complete message acknowledges the receipt of packet data forwarded from the old SGSN, the packet data shall be rejected by the new SGSN.
In summary, there are two levels of location management of GPRS: Narrow roaming management performed by the SGSN allows storing information about the current routing area or current cell of the mobile station and wide roaming management allows storing information about the current SGSN of the mobile station in the HLR, VLR, GGSN.
2.6.3. Network connection procedure (Attach)
When a GPRS subscriber wants to send or receive data, it performs the network access procedure. The GPRS network access operation is to inform the network about the presence of the MS on the network. During the network access procedure (it can be GPRS access, IMSI access or a combination of GPRS/IMSI). After the MS performs the network access operation, the MS switches to the ready state and the mobile management frame is established in the MS and SGSN. After the MS has been connected to the network, it can also receive SMS via GPRS and make calls via GSN. The mobile station can initiate a packet data protocol frame (PDP context) which is a mandatory operation when the MS wants to receive and send GPRS data.
For subscribers using both circuit-switched and packet-switched services, both GPRS and IMSI network entry procedures can be combined. The figure below illustrates the GPRS network entry procedure from the MS.
GGSN
HLR
MS
BSS SGSN
1. Network entry requirements
2. Authentication
4. Accept network entry
5. Complete network login
2. Authentication
3. Update location
Figure 2 – 15: Description of the network entry procedure from MS
1. The mobile station (MS) initiates the network attachment procedure by transmitting an association request message to the SGSN. This association request message contains information about the mobile station.
2. If the Mobility Management Frame (MMC) already exists on the network, the authentication function must be used. The authentication function is also used in case the maximum number of mobile stations that have joined the network has been reached. After the temporary mobile subscriber identity P-TIMSI has been assigned, if the network uses encryption, the encryption mode will be selected.
3. If the mobile station changes its SGSN after joining the GPRS network, or in the case of the mobile station's first network entry. The SGSN will notify the HLR to update the location of the mobile station. The HLR will then notify the old SGSN if the mobile station changes its SGSN.
4. The SGSN selects the radio channel and sends an admission message to the mobile station, the P-TIMSI number will be sent along if the SGSN assigns a new P-TIMSI number.
5. If the P-TIMSI number changes, the mobile station shall notify the SGSN of the receipt of the P-TIMSI number by means of a network attachment completion message.
2.6.4. Network disconnection procedure
Similar to network attachment, there are three basic types of network detachment: IMSI detachment, GPRS detachment and combined GPRS and IMSI detachment. Combined GPRS and IMSI detachment can only be initiated by the MS.
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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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2.6.4.1. Description of the procedure for detaching from the mobile station Below is a description of the procedure for detaching from the mobile station.
MS BSS SGSN
1. Network requirements
3. Accept network failure
2. Request for scratch data frame (P
2. Answer the question
Open frame of CD/SD card (
Mandatory procedure
Optional or conditional
Figure 2 – 16: Procedure to leave GPRS network from mobile station
1. The mobile station detaches by sending a detach request to the current SGSN. The detach request contains information about the type of detachment (GPRS detach, IMSI or connection).
GPRS/IMSI combination) and information on whether the network disconnection was due to mobile phone shutdown.
2. If the network disconnect type is GPRS, the packet data protocol frame that was initiated at the GGSN shall be deleted by the SGSN by sending a packet data protocol frame deletion request to the GGSN. The GGSN shall reply with a packet data protocol frame deletion request reply message.
3. If the detachment is due to mobile station shutdown, the SGSN will send a detachment acceptance message to the mobile station.
2.6.4.2. Procedure for leaving the network from HLR
The Home Location Register (HLR) uses the HLR detach procedure to operate the network, it can perform the request to delete the mobile management frame and packet data protocol frame (MM frame and PDP frame) of the subscriber at the SGSN. The GPRS detach procedure from the HLR is described as below:
2. leave request
4. Accept to leave the network
1. Delete
MS
BSS
SGSN
GGSN
HLR
network
h position
3. Clear packet data protocol frame
3. Request to delete packet data protocol frame
5. Confirm deletion of location
Mandatory Procedure
Optional or conditional
Figure 2 – 17: Procedure to leave GPRS network from HLR
1. If the HLR wants to immediately delete the mobile management frames and packet data protocol frames of the subscriber at the SGSN. The HLR will send a delete location message to the SGSN.
2. The SGSN notifies the mobile station that the mobile station has left the network by sending a Leave Request message to the mobile station.
3. The packet data protocol frame that has been initiated and is active at the mobile station's SGSN shall be deleted by the packet data protocol frame deletion request message from the SGSN to the GGSN. The GGSN shall confirm the deletion of the packet data protocol frame by sending a packet data protocol frame deletion request reply message.
4. The mobile station sends a detachment acceptance response message to the SGSN immediately after receiving the detachment request.
5. The SGSN confirms the deletion of mobility management frames and packet data protocol frames with a location deletion acknowledgement message.
2.6.5. Enable Packet Data Protocol – PDP
After the network attachment procedure, the MS performs the packet data protocol activation procedure. Normally the MS requests the network to activate a PDP with a certain quality of service. However, the PDP can also be activated by the network requesting the MS. During the PDP activation process, the router at the GGSN is also activated. Routing between the SGSN and the GGSN is performed by activating tunneling between the SGSN and the GGSN. PDPs can be activated for fixed or dynamic addresses. After the network attachment and PDP activation, the MS can send and receive point-to-point or point-to-multipoint information.
2.7. Transmission characteristics and power output correction
Messages on the air interface are transmitted over radio blocks. Each radio block is composed of four interleaved bursts. The radio block consists of a MAC header, a body containing data or signaling information, and a block checksum. There are four different encoding methods:
CS – 1 has a speed of 9.05 kbps/1 timeslot, data rate of 8 kbps/1 timeslot.
CS – 2 has a speed of 13.4 kbps/1 timeslot, data rate of 12 kbps/1 timeslot.
CS – 3 has a speed of 15.6 kbps/1 timeslot, data rate of 14.4 kbps/1 timeslot.
CS – 4 has a speed of 21.4 kbps/1 timeslot, data rate of 20 kbps/1 timeslot.
The decision on which coding method to use depends on the network conditions, or more specifically the quality of the radio link between the mobile station and the base station. If the radio link quality is poor, there is a lot of interference, and the reliability is not high, the network will use


![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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