An Analysis of Error Handling Techniques in Voice over IP

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1 An Analysis of Error Handling Techniques in Voice over IP Martin John Lipka ABSTRACT The use of Voice over IP (VoIP) has been growing in popularity, but unlike its wired circuit-switched telephone network predecessor, this technology runs over a packet-switched network. It is therefore vulnerable to changes in underlying network conditions such as network delays and packet loss due to congestion. Many believe that VoIP is the future replacement for today s wired telephony and that it will be widely adopted. In order for this technology to compete with circuit-switched telephone networks, the underlying assumption throughout this paper is that if an error occurs in the transmission stream, it needs to be overcome or handled in some way. This paper focuses on which techniques may help VoIP reach a quality of service (QoS) level that is comparable to circuit-switched telephone networks by focusing on their ability to minimize both packet loss and jitter while increasing the overall sound quality of the voice transmission. The metrics used consist of performance-oriented criteria that ultimately affect the end user s perceived QoS. Thus, some of the error recovery and concealment techniques most commonly used today are introduced, their features analyzed, and an assessment made of their effectiveness. Keywords Voice over IP (VoIP), Error concealment, Error recovery 1. INTRODUCTION As the Internet grows in popularity, new technologies are discovered and become more prevalent with time, usually starting in the business arena and slowly working their way down to the consumer market. One of these new technologies focuses on the transmission of voice over the Internet, enabling two or more parties to communicate with each other. In the last few years, voice over IP or VoIP, has grown immensely in its popularity. Businesses for quite some time have realized the features and cost savings VoIP provides them; now consumers are also realizing these same benefits. VoIP is the likely candidate to replace today s method of how we transmit voice over highly reliable, wired telecommunication networks. The Internet, however, and the underlying protocols may be unreliable and are considered best effort over a packet-switched network. With the advent of voice over IP and its growing popularity to the home user, many things should be considered regarding errors in the voice transmission. These include what should be done when an error within the audio transmission is discovered and how, or whether, one should recover from such an error. Furthermore, circuit-switched networks like the ones used by phone companies today provide a high level of service and quality mainly because a point-to-point connection is established between both parties before a conversation takes place. Errors such as delay or static in the circuit-switched voice environment occur rarely and are usually unnoticeable. This, however, is not the case in voice over IP. The VoIP technology runs over an unreliable medium (the Internet) and therefore it is very susceptible to disruption from increased network traffic, congestion, dropped packets, or delays within the Internet, which could lead to errors within the audio transmission resulting in silence, jitter, and a decrease in the overall sound quality. This paper examines whether some of the error recovery and concealment techniques used in VoIP and other related technologies such as MP3 streaming are in fact viable solutions to attaining the same level of quality of service currently present in wired telecommunication networks under the assumption that VoIP is a type of controlled-load network service. As described in RFC 2211, a controlled-load network has no specific guarantees, but it does offer a constant level of service [9]. RFC 2212 discusses the specifications necessary to guarantee a certain level of quality of service [5], and after considering its recommendations the author concludes that VoIP QoS can be grouped into two main categories, namely network performance QoS and end-user QoS. As the network performance QoS increases, the end-user QoS increases as well. What follows is a discussion of voice over IP with a focus on the issues surrounding the transmission of the voice data and ultimately an evaluation is presented on how some of the VoIP error recovery and concealment techniques are effective with respect to specific criteria in the above two categories. The criteria considered when evaluating the error recovery and concealment techniques for VoIP include: (1) for end-user QoS, ability to increase the overall sound quality and (2) for network performance QoS, ability to handle errors due to packet loss and jitter caused by network delays. These are key for VoIP to match the QoS found in today s circuit-switched telephone networks where there is little to no packet loss or jitter and the overall sound quality is quite high. The error recovery and concealment techniques are examined in terms of three characteristics in order to determine their overall effectiveness in improving the QoS categories specified earlier: ability to handle packet loss, ability to handle jitter, and ability to improve overall sound quality. We define packet loss as the absence of an expected packet at the receiver. Jitter within the audio stream is the variation in time between arriving voice packets at the receiver, resulting from variable delay values such as those incurred from packetization, transmission, and propagation delays. These two criteria were chosen because they have a direct impact on and are vital to the third characteristic, overall sound quality. In contrast, circuitswitched telephone networks offer excellent sound quality with little to no loss of data or jitter mainly because a dedicated connection is established before transmission of data (voice). SB3-T3-1

2 2. VOICE COMMUNICATION OVER the INTERNET Speech is an analog signal whose bandwidth does not exceed 4 khz [2]. To transmit voice over the Internet, the sound needs to be converted by the sender from analog to digital, and then back to analog on the receiving end. The conversion uses certain compression or encoding algorithms commonly referred to as encoding schemes. Typical encoding schemes include: Pulse Code Modulation (PCM): This includes technology such as G.711 that is capable of producing a digital signal at 64 kb/s [2]. It contains built-in packet loss concealment techniques but does not contain techniques implementing compression [1]. Code Excited Linear Predictive (CELP): CELP technologies such as G.729 and G are capable of producing digital signals at 8 kb/s and 6.4 kb/s respectively [2] and contain built-in packet loss concealment techniques [1]. CELP technologies are attractive because they produce signals at lower data rates at the expense of potentially reduced quality, additional complexity and encoding delays [2]. Although the data rates that CELP provides are quite low and attractive, further improvement is possible by noticing that speech is not a continuous stream of audio, but is in fact composed of silence and sound. This offers the ability to encode the audio only when there is sound and not to encode the silence, which is known as Voice-Activated Detection (VAD) [2]. The capability this provides is very beneficial to the transmission of voice because it requires less available bandwidth when compared to encoding both silence and sound. The transmission of real-time information such as voice over the Internet is typically made possible by using the Real-Time Transport Protocol (RTP), which runs over the User Datagram Protocol (UDP). RTP provides the packet sequence information so that packets are kept in order and provides time stamping to help avoid jitter, while UDP provides the end-to-end delivery but is considered unreliable. That is, there is no guarantee that data will reach the intended receiver or the packets will arrive in order, because successful transmission is dependent on a variety of network conditions such as congestion and connectivity. With the increased popularity of VoIP and the many features it brings, customers will expect it to have the same type of voice quality and reliability that public switched telephone networks offer today [2]. The Internet as a whole is an ever-evolving technology and there are many people and corporations trying to improve its quality and reliability in regards to VoIP. Many of the Internet Service Providers (ISPs) are very capable of offering VoIP technology, but there are still many network paths that can cause poor performance [2]. In fact, a large number of Internet paths have high delay and high delay variability [2], which are very unfavorable to VoIP. The quality the end user will experience can be measured by the network loss and delay within the network. Typically, low delay and low delay variability lead to excellent VoIP performance [2]. Both factors are very dependent on the amount of traffic on the network. When the amount of traffic increases, congestion may cause packet delay or even lost packets. In VoIP, this translates to errors in the audio stream. It is then important to overcome these errors to improve the perceived quality and be able to offer the same level quality of service found in circuit-switched telephone networks. To overcome network delay and offer a continuous audio stream, playout buffers are typically used at the receiver. These offer some error concealment by discarding any packets that arrive after their successor packets have been played, but the end user may still experience jitter and an audio stream that is below the quality found in public switched telephone networks. 3. QUALITY of VOICE TRANSMISSION The quality of a voice transmission is dependent on a variety of factors. As discussed, the type of encoding does play a significant role in determining the quality one is likely to experience. While encoding is important, the underlying network conditions have an even larger impact on what is likely to be the overall quality of the voice received. Things that can affect the level of quality may even include the priority assigned to packets carrying voice. If these packets have a higher priority than other packets on the network, then they are more likely to reach their destination. The opposite holds true if the packets have a lower priority than other packets on the network. Furthermore, as network congestion increases and packets are lost, the receiver of the voice transmission will experience speech clipping [3]. The Resource Reservation Protocol (RSVP), RFC 2750, is widely used in real-time transmission of both voice and video [6]. It requires that resources be allocated or reserved by the network prior to transmission. This might not be a viable option in VoIP where large-scale deployments are desired, specifically because as the number of users increases, the demand for resources increases, which translates into more needed equipment and higher costs. If the unreliability of UDP is seen as an issue for VoIP, TCP could be considered instead. However, TCP is not entirely appropriate for VoIP. Real-time communications require a low delay, which is not something the TCP retransmission techniques promote [3]. When trying to measure quality of voice communication, the Mean Opinion Score (MOS) is widely used. It offers the ability to determine the quality of the VoIP transmission by having a variety of people listen to audio clips and then rate them from 5 to 1 with 5 meaning Excellent, 4 Good, 3 Fair, 2 Poor, and 1 meaning bad [1]. On the Internet the most common type of errors in the transmission of data results from bursty loss. When rating an audio clip with bursty loss, listeners assign lower MOS values than for random loss [1]. On the other hand, as random loss increases and becomes more frequent, the difference between bursty loss and random loss becomes less [1]. In addition, environmental characteristics such as echo or noisy surroundings may also affect the quality, as can transmission noise experienced over copper wire. Even if packets are not lost, delay can have an effect on perceived voice quality. Packetization delay is generally uniform, but transmission delay depends on the length of path traveled. Additional delay due to congestion in the network can cause even greater variation in delay packet by packet. SB3-T3-2

3 Encoding and decoding between analog and digital signals also adds delay, which could be variable if compression is performed. These delay variations may well cause a receiver to drop packets that arrive late, compounding the jitter with additional packet loss. 4. PLAYOUT SCHEDULING TECHNIQUES Playout scheduling determines how the packets received should be played for a listening end user. Packets may arrive out of order, arrive late, or not arrive at all. It is the responsibility of the receiving end to determine the best way to handle these situations. The two types of playout scheduling schemes are: Fixed - Fixed scheduling schemes play the voice stream in such a way as to make the end-to-end delay the same for all arriving packets [2]. If the buffering delay is small, this technique improves interactivity while increasing loss due to late packet arrival (i.e., the time spent waiting for a late packet is small) [2]. Large delay buffers, on the other hand, decrease packet loss but also decrease the interactivity [2]. Therefore, it is important to examine closely the delay a packet is likely to experience in the network before choosing a fixed end-to-end delay value for playout. Adaptive - Adaptive playout scheduling schemes constantly examine and monitor the network delay [2]. The playout of the packets is adjusted to compensate for changes in delay in order to optimize the quality of the voice reception. Timestamps on each packet are used, as well as the differences in timestamps between successive speech frames, at the receiving side [3] to help determine the appropriate playout of the packets to compensate for any experienced end-toend variability in delay, that is, the jitter. 5. CLASSIFICATION of ERROR HANDLING TECHNIQUES The error concealment and recovery techniques, also known as error handling schemes, can be classified into two main categories: Reactive This is sometimes called an active errorhandling technique because it focuses on handling the error right when it has been detected. These techniques can utilize feedback with retransmission and are either sender- or receiver-based [6]. Proactive - Proactive error handling or passive error handling is the opposite of reactive. It focuses on error avoidance, trying to foresee and correct for errors before the actual transmission takes place. The most common form is Forward Error Correction (FEC), which can be either sender- or receiver-based [6]. Table 1 categorizes the different types of error handling techniques related to VoIP, specifying whether they are reactive or proactive, and if they are sender- or receiver-based or possibly both. Section 6 discusses these in further detail. Table 1. Error handling techniques and their attributes Technique Classification Type Beat Pattern Error Concealment Reactive Receiver-based Intra-Flow Loss Recovery and Control Proactive Both Packet Loss Concealment Reactive Receiver-based Muting Reactive Receiver-based Interpolation Reactive Receiver-based Retransmission Reactive Sender-based Feedback Reactive Receiver-based Interleaving Proactive Sender-based Repetition Reactive Receiver-based Error Spreading Reactive Receiver-based Forward Error Correction Proactive Sender-based Low Bit-Rate Redundancy Proactive Sender-based 6. ERROR HANDLING TECHNIQUES VoIP runs over the Internet and because the Internet is inherently unreliable, errors are likely, due to a variety of factors and delays within the network. Once an error is experienced, it is important for quality purposes to use a technique to recover from or conceal this error. What follows is a description and analysis of the techniques and methods most commonly used and most applicable to error recovery and concealment in VoIP and audio transmission across the Internet in general. Beat Pattern Error Concealment - One of the most common and popular formats for streaming music on the Internet is the MP3 format. Music, unlike voice, has a very distinct rhythm. Repetition of these rhythms is also very frequent. By considering these two aspects, one method of concealing the errors experienced in a MP3 audio stream consists of beat pattern error concealment [8]. The process of concealing the error is actually very straightforward. It consists of looking at the structure of the music and noting its beats, specifically those that repeat frequently in the audio. An error in the audio stream typically results in a lost beat. Beat pattern error concealment tries to cover up this error by replacing the lost beat with a beat that was already played [8]. This technique, although very viable and appropriate to error concealment in music, is not applicable to voice and VoIP. Voice does not follow a distinct pattern or repetition, at least to the extent that music does, and for this reason, other alternative techniques are more applicable to concealing errors in the transmission of voice. Intra-Flow Loss Recovery and Control - The actual stream of audio from sender to receiver, which in this SB3-T3-3

4 case is voice, is known as a flow. Intra-flow loss recovery and control is concerned with correcting the errors as the audio stream travels from the sender to the receiver. Within VoIP, this technique is known to increase the perceptual quality an end user experiences [4]. This approach is based on the premise that a short burst of lost packets or data makes it difficult to conceal the error at the receiver, which leads to interruptions within the audio stream. Remembering that voice conversations are made up of periods consisting of both silence and sound, we conclude that some packets are in fact more important than others are. That is, those packets containing sound should be considered more important than those containing silence. If a packet containing silence is lost, the impact on the quality of the stream is lower than if the packet contains audio. This leads to a practice of marking those packets considered more important after looking at how sensitive they are to loss. Intraflow loss recovery focuses on marking those packets within the flow that are sensitive to loss on the sender s side and those sensitive to loss concealment on the receiver s side [4]. By marking such packets, both the sender and receiver can determine which packets can be lost with the least amount of impact on the overall quality of the stream. The differentiation of those packets that are sensitive to loss can be grouped into two categories, temporal sensitivity, and sensitivity due to Application Data Unit (ADU) heterogeneity [4]. Temporal Sensitivity is a form of time dependence in which all packets are both dependent on each other and on time as well. A packet should only be decoded if the one preceding it has been decoded [4]. This ensures that the receiver receives the transmission in the order it was sent and makes sure that the audio stream is not played out of order due to late or early arrival of packets. However, if one of these packets is lost, this is considered an isolated packet loss and not a loss that occurs in bursts [4]. An Application Data Unit (ADU) is a unit of data emitted by a source coder [4]. Heterogeneity fits with the concept that some ADUs or packets are more important to determining the perceived quality of the voice transmission experienced by end users and are thus more sensitive to loss. If for example, a packet containing silence is lost, the impact on the perceived quality is minimal compared to a lost packet containing sound. Therefore, it is important to try to identify those packets considered more important so that error recovery can be performed correctly if the packet is lost. The main problem with this technique is that the adaptation of the sender s bitrate to the current network congestion state as an intra-flow QoS scheme is difficult to apply to voice [4]. When looking at streaming audio (MP3 music), in general we can conclude that it takes fewer bits to encode a voice conversation than the MP3. However, when a packet is lost at the receiving end, the cost to transmit feedback to the sender that the packet was lost is higher because VoIP is a real-time communication technology. Furthermore, when an error is identified and recovery initiated, significant amounts of time may pass making these lost packets irrelevant because new ones have already arrived. Packet Loss Concealment (PLC) This consists of creating a replacement for the lost packet by creating a packet similar to what has already been played. This technique is used on the receiver s end and it is believed that A good PLC algorithm can greatly improve the perceived quality with no bandwidth overhead [1]. Muting When an error such as a lost or delayed packet is identified on the receiver s side, a simple method of recovering from the error is to insert silence or noise into the decoded audio stream in place of the missing bits [2]. This is the simplest of the error concealment techniques in terms of computational requirements, but the result of inserting silence into the stream is not necessarily desirable to end users, thus lowering the perceived quality of service. Interpolation Once the loss of a packet is discovered, recovery begins with examining the area of the lost packet in relation to other packets around it. By looking at the characteristics of the sound (speech), and examining its waveform in the area of the lost packet, either a replacement packet can be generated to fill the gap or a previously played packet that most closely fits the spot where the error is might be used. This technique requires both computation and buffering or storing of previously played packets. Retransmission One of the most common ways to recover from lost or corrupt data due to transmission on the Internet is retransmission. Although this is a very appealing and widely used technique for static data, it is not practical for VoIP because of its realtime nature [2]. If retransmission is desired, it should only be used when the end-to-end delay between the sender and receiver is very small. Feedback Notifying the sender to change the rate at which the audio stream is being sent is a feedback technique. One may increase or decrease the data stream speed, change the rate at which the audio is encoded, or notify the sender to switch to a different type of encoder [2]. Interleaving This technique focuses on separating the voice signal into several frames where each frame is just a section of the original audio stream [7]. Just before transition, the order of these frames is scrambled, each is placed into a packet, and the results rearranged back to their original order at the receiver s end [2]. In other words, the frames are interleaved over multiple packets, thus spreading the probability of an error across multiple sections in the audio stream. The goal is to optimize the order in which the packets are sent but this requires CPU calculations in SB3-T3-4

5 order to separate the audio into frames and to place them into separate packets. Therefore, this technique is not entirely applicable to VoIP. It may add additional delay, which is something that needs to be avoided when transmitting real-time data such as VoIP. It is however a good option for streaming prerecorded audio, especially to small mobile devices or for streaming MP3 audio in general [7] as long as it isn t a real-time audio recording such as VoIP. Repetition Another simple form of error recovery and concealment is to repeat the audio that was last played in the packet arriving just before the error was encountered [1]. For real-time data such as audio streaming, this is acceptable if the lost section of audio is smaller than approximately 20ms [8]. On the other hand, this technique is not entirely acceptable for voice conversations within VoIP because depending on the end user s ability to discern imperfections, it might be irritating if the lost section was in excess of 20ms. If the repeated sections are 20ms or less, however, then this repetition would be virtually unnoticeable by the end users because the error would result in a small stutter within the audio stream. Nevertheless, as parts of the conversation are repeated, the level of quality is reduced even if the end user is unable to notice it. Error Spreading This technique tries to decrease the impact errors have on the quality of the stream by spreading the error across a larger portion of the audio stream than just the general area where the error occurred. This increases the perceived quality of the stream [6]. It works by making the error less noticeable to the user through permuting the input sequence of packets before transmission and unscrambling them on the receiving end [6]. Forward Error Correction (FEC) Forward error correction is the most popular form of error correction on the Internet. It works on an end-to-end basis to recover from loss due to an error while still meeting the real-time delay constraints [4]. This technique works by transmitting extra, redundant information in the same packet along with the original data in order to have enough information to reconstruct the packet if any of its sections are damaged in transmission. It is therefore attractive for transmitting audio in VoIP, but there are some disadvantages. FEC recreates the lost packets and it can reduce the effect of errors but not eliminate them [1]. Furthermore, the amount of redundant information being sent needs to be adaptive to avoid taking bandwidth away from other flows [4]. In other words, only the necessary amount of redundant information to recreate the lost packet should be transmitted. In addition, the packet reconstruction process may introduce additional delay at the receiver when trying to correct jitter [4] and as the size of the redundant FEC data increases, it may increase the overall end-to-end delay [1]. Low Bit-Rate Redundancy (LBR) LBR is considered an alternative to FEC in which a copy of the audio packet is always sent but at a lower quality [1]. Therefore, the lost packet containing the original higher-quality audio can be substituted simply by using the copy. An unrecoverable error might be experienced when both the original and the copy are lost. Nevertheless, MOS tests indicate, FEC is much preferred over LBR [1]. 7. ANALYSIS With VoIP being a technology that runs over the Internet, errors should be expected. The Internet and its underlying protocols consist of a best-effort delivery system. There is no guarantee that information will reach the intended receiver and this may present a problem for VoIP and the quality of service it can offer. Error recovery and concealment techniques try to improve this quality when an error is encountered. Error concealment focuses on hiding (concealing) the error from the end user so as to offer the perception that the QoS level hasn t deteriorated even though the actual audio transmission has been affected by network conditions. Error recovery focuses on trying to fix or compensate for the error in order to maintain or even improve the perceived QoS for the end user. Both concealment and recovery are vital to quality and should be used for VoIP. There are many methods and techniques for recovering from errors experienced in the transmission of voice using VoIP, but no one technology is best. As depicted in Table 2, Forward Error Correction (FEC) seems to be the most favorable, but it alone is not sufficient. An enhanced error recovery and concealment technique consists of a combination of more than one technique and the decision of which combination is best can only be determined by considering a variety of factors, including the type of error, its size, and the state of the network. The ideal implementation of this combination however, is one that is capable of dynamically adapting to the constant changes in the network, and doing so in real time. Dynamic adjustment of a combination is inherently a CPU-intensive process. CPU time and computer resources at either the sender or receiver (or possibly both) are needed to calculate the state of the network and adapt accordingly. This may introduce additional delays into the audio transmission from the overhead of the dynamic process. Therefore, a static combination is more appropriate for VoIP. In order to minimize any induced delays resulting from a dynamically chosen combination, we propose a static combination be chosen ahead of time, independent of any changing network conditions. The scores in Table 2 suggest that a combination of FEC, Error Spreading, and Feedback would provide the most powerful way to conceal or recover from most if not all of the errors. Such a combination would certainly do a better job at concealing and recovering from errors than any of the techniques used alone. Both Feedback and Forward Error Correction provide mechanisms for handling or reducing the likelihood of packet loss, error spreading helps reduce the impact of network delays that induce jitter into the audio stream, and all three help to improve the overall sound quality. Determining whether this is the best possible combination and how much better it would be than other combinations or individual techniques, requires a detailed investigation of both network congestion and computational requirements which are beyond the scope of this paper. SB3-T3-5

6 Table 2. Error handling techniques and their effectiveness in improving QoS Technique Beat Pattern Error Concealment Intra-Flow Loss Recovery and Control Packet Loss Concealment Jitter Packet Loss Overall Sound Quality Score Muting Interpolation Retransmission Feedback Interleaving Repetition Error Spreading Forward Error Correction Low Bit-Rate Redundancy Key: - poor (0), + good (1), + + better (2), best (3) 8. FUTURE WORK The area of VoIP technology is becoming more popular each year as businesses and consumers understand the features and realize the benefits it offers. VoIP can also result in significant cost savings when compared to traditional public switched telephone networks. Besides the error recovery and concealment techniques described in this paper, future areas to be examined include reliability, security, scalability, automatic failover, and fault detection within VoIP as compared with circuit-switched networks. These could be inspected and researched in order to determine what additional steps would make VoIP as good as, if not better than, wired circuit-switched telephony. 9. REFERENCES [1] Jiang, W; Schulzrinne, H; Comparison and Optimization of Packet Loss Repair Methods on VoIP Perceived Quality Under Bursty Loss, International Workshop on Network and Operating System Support for Digital Audio and Video archive Proceedings of the 12th international workshop on Network and operating systems support for digital audio and video, Miami, Florida, 2002, pp [2] Markopoulou, A; Tobagi, F; Karam, M; Assessing the Quality of Voice Communications Over Internet Backbones, IEEE/ACM Transactions on Networking (TON) archive, Vol. 11, Issue 5, 2003, pp [3] Matta, J; Pepin, C; Lashkari, K; Jain, R; A Source and Channel Rate Adaptation Algorithm for AMR in VoIP Using the Emodel, International Workshop on Network and Operating System Support for Digital Audio and Video archive Proceedings of the 13 th international workshop on Network and operating systems support for digital audio and video, Monterey, California, 2003, pp [4] Sanneck, H; Long Lee, N; Wolisz, A; Carlie, G; Intraflow Loss Recovery and Control for VoIP, International Multimedia Conference archive Proceedings of the ninth ACM international conference on Multimedia, Ottawa, Canada, 2001, pp [5] Shenker, S; Partridge, C; Guerin, R; "Specification of Guaranteed Quality of Service" RFC 2212, September [6] Varadarajan, S; Ngo, H; Srivastava, J; Error Spreading: A Perception-Driven Approach to Handling Error in Continuous Media Streaming, IEEE/ACM Transactions on Networking archive, Vol. 10, Issue 1, 2002, pp [7] Wang, Y; Huang, W; Korhonen, J; A Framework for Robust and Scalable Audio Streaming, International Multimedia Conference archive Proceedings of the 12th annual ACM international conference on Multimedia, New York, New York, 2004, pp [8] Wang, Y; Vilermo, M; A Compressed Domain Beat Detector Using MP3 Audio Bitstreams, International Multimedia Conference Proceedings of the ninth ACM international conference on Multimedia, Ottawa, Canada, 2001, pp [9] Wroclawski, J; "Specification of the Controlled-Load Network Element Service" RFC 2211, September SB3-T3-6

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