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Implementation of Video Transfer with TCP-friendly Rate Control Protocol
`
`Masaki Miyabayashi, Naoki Wakamiya, Masayuki Murata, Hideo Miyahara
`Department of Informatics and Mathematical Science,
`Graduate School of Engineering Science, Osaka University, Japan
`1-3 Machikaneyama, Toyonaka, Osaka 560-8531, Japan
`Tel: +81-6-6850-6588, Fax: +81-6-6850-6589
`E-mail: {miyabays, wakamiya, murata, miyahara}@ics.es.osaka-u.ac.jp
`
`Abstract: As the use of real-time multimedia applications
`increases, a considerable amount of “greedy” UDP traffic
`would easily dominate network bandwidth and packet loss.
`As a result, bandwidth available to TCP connections is op-
`pressed and their performance extremely deteriorates. In or-
`der that both TCP and UDP sessions fairly co-exist in the
`Internet, it is vital that we consider the fairness among both
`protocols. In this work, we implement a “TCP-friendly” rate
`control mechanism suitable to video applications and con-
`sider its applicability to a real system through observation of
`the video quality at the receiver and the connection state. It
`is shown that we can achieve high-quality and stable video
`transfer fairly sharing the network bandwidth with TCP by
`applying our rate control at a control interval of 32 times as
`long as RTT.
`
`Introduction
`1
`Since the current Internet does not provide QoS (Quality
`of Service) guarantee mechanisms, each application chooses
`the preferable transport protocol to achieve the required QoS
`level. For example, traditional data applications such as http,
`ftp, telnet employ TCP which accomplishes the loss-
`free data transfer by means of window-based flow control
`and retransmission mechanisms. On the other hand, real-
`time multimedia applications such as video conferencing pre-
`fer UDP in order to avoid the unacceptable delay introduced
`by packet retransmissions. Against the network congestion
`TCP throttles its transmission rate, whereas UDP does not
`have such control mechanisms. As real-time multimedia ap-
`plications increase, a considerable amount of “greedy” UDP
`traffic would dominate network bandwidth. As a result, the
`available bandwidth to TCP connections is oppressed and
`their performance extremely deteriorates.
`In order that both TCP and UDP sessions fairly co-
`exist in the Internet, it is meaningful to consider the fairness
`among protocols. In recent years, several researches have
`been focused on the investigation of the “TCP-friendly” rate
`control [1-7]. “TCP-friendly” is defined as “a non-TCP con-
`nection should receive the same share of bandwidth as a TCP
`connection if they traverse the same path” [4]. With the rate
`control based on the concept of ”TCP-friendly”, a UDP con-
`nection achieves a fair share of the network bandwidth with
`a TCP connection on the same path by regulating its sending
`rate according to the network condition.
`Some researches on TCP-friendly rate control have been
`devoted to the applicability to real-time MPEG-2 video com-
`munications, and the effective control mechanisms have been
`
`control interval
`2−iI
`sending rate
`2−ir
`
`1−iI
`1−ir
`
`sender
`
`iI
`
`Loss estimation
`ir and
`determine
`iI
`ir
`
`lost
`
`receiver ACK
`Figure 1: TFRCP video transfer
`
`1+iI
`
`1+ir
`
`lost
`
`time
`
`proposed [6, 7]. However, the effectiveness of those mech-
`anisms is evaluated through simulation experiments assum-
`ing the ideal network system environment. That is, they do
`not take into account several factors which would affect the
`performance of the rate control. Those include the fluctu-
`ation of control interval or the quality degradation during
`playback of a video sequence. In this work, we implement
`TCP-friendly rate control protocol suitable to video appli-
`cations (called MPEG-TFRCP), which is proposed by our
`research group. We demonstrate its applicability to a real
`system through evaluation of the perceived video quality and
`observation of the traffic on the link.
`The paper is organized as follows. In Section 2, we
`introduce MPEG-TFRCP and show some results of our real
`system experiments.
`In Section 3, we consider the mech-
`anism to improve our proposed MPEG-TFRCP algorithm.
`Finally, we summarize our paper and outline our future work
`in Section 4.
`
`2 MPEG-TFRCP
`In this section, we introduce MPEG-TFRCP suitable for video
`transfer, which is proposed by our research group [6, 7]. In
`order to transmit a video sequence with high and stable qual-
`ity while fairly sharing the network bandwidth with TCP, our
`MPEG-TFRCP behaves as illustrated in Fig. 1; at the end of
`the control interval i − 1 whose duration is Ii−1, the sender
`estimates the network condition from information gathered
`within the interval, then derives the throughput of a TCP ses-
`sion rT CP , and finally regulates its sending rate ri for the
`next interval i.
`
`2.1 MPEG-TFRCP mechanism
`In [6, 7], the MPEG-TFRCP mechanism achieves fairness
`among TCP and UDP connections by adjusting the sending
`rate to the estimated TCP throughput at the regular interval
`of 32 times as long as RTT. The target rate of interval i (de-
`noted as ri ), is determined as
`
`VIMEO/IAC EXHIBIT 1024
`VIMEO ET AL., v. BT, IPR2019-00833
`
`Page 1 of 4
`
`

`

`60
`
`60
`
`UDP
`TCP
`
`0
`
`10
`
`30
`20
`GoP times
`Figure 4: Rate variation (UDP vs. TCP)
`
`40
`
`50
`
`MPEG-TFRCP
`TCP
`
`10
`
`02468
`
`10
`
`02468
`
`0
`
`10
`
`40
`
`50
`
`30
`20
`GoP times
`Figure 5: Rate variation (MPEG-TFRCP)
`
`rate [Mbps]
`
`rate [Mbps]
`
`reported number and the round trip time by observing the
`timestamp.
`
`2.3 Evaluation of MPEG-TFRCP
`When both TCP and UDP sessions co-exist in the network,
`the performance of TCP connection is heavily affected by
`the UDP traffic as shown in Fig. 4. In the experiment, both
`sessions transfer the MPEG-2 video data of the average rate
`5 Mbps. The experiment time on an x-axis is expressed by
`the GoP time of MPEG-2 which is equal to 1.001 sec (GoP
`size/frame rate = 30/29.97). Y-axis express the actual rate
`measured on the link with tcpdump. In this experimental
`environment, we observe that RTT is about 10 ms.
`On the contrary to Fig. 4, Fig. 5 shows the experi-
`mental result where TCP and MPEG-TFRCP sessions com-
`pete for the bandwidth. It can be observed that the MPEG-
`TFRCP connection regulates its data sending rate during the
`session. However, Fig. 5 is quite different from results in [6,
`7] owing to the large packet loss probability and the unstable
`RTT introduced by the processing delay at the PC router and
`the end systems. For instance, Fig. 6 shows the packet loss
`probability p (”loss”), the target rate ri (”target rate”) esti-
`mated by Eqs. (1) and (2) and the average of the actual rate
`(”video rate”) of MPEG-TFRCP. Here we should note that
`there are some durations where the video rate is higher than
`the target rate because video data we employed in the exper-
`iments ranges from 1.138 Mbps to 27.260 Mbps. In those
`cases, the sender first sends as much data as possible, then
`
`PC
`
`PentiumII
` 200Mhz
`
`10Base−T
`
`10Base−T
`
`router
`
`PC
`
`TCP
`PentiumII
` 233Mhz
`
`Senders
`
`HUB
`
`10Base−T
`
`10Base−T
`
`10Base−T
`
`PentiumII
` 233Mhz
`
`HUB
`
`10Base−T
`
`PC
`
`PC
`
`Receivers
`
`{ TCP
`
`}
`
`UDP/MPEG−TFRCP
`PentiumIII
` 700Mhz
`
`UDP/MPEG−TFRCP
`PentiumIII
` 700Mhz
`
`Figure 2: System configuration
`
`Camera
`
`Hard
`Disk
`
`Encoder
`
`Quantizer
` Scale
`
` QoS
` Manager
`
`Trans−
` mitter
`
`RTP
`
`video data
`
`RTP
`
`UDP
`
`Sender Report
`
`UDP
`
`Receiver
`
`target rate
`
`Estimator RTCP
`
`RTCP
`
`Receiver Report
`
`Decoder
`
`Monitor
`
`Sender
`Figure 3: MPEG-TFRCP sender & receiver
`
`Receiver
`
`
`
`
`ri =
`
`rT CP ≈
`
`(cid:5)
`
`M T U
`3 + T0 min(1, 3
`
`2p
`
`(cid:5)
`
`RT T
`
`2 × ri−1,
`
`,
`
`3p
`
`8 )p(1 + 32p2)
`(1)
`if p > 0
`
`if p = 0
`
`(2)
`
`where MTU stands for the maximum transfer unit size, p is
`the packet loss probability, RTT and T0 are for the round trip
`time and the retransmission timeout, respectively. The net-
`work condition (conjectured by RTT and packet loss proba-
`bility) is estimated from the feedback information obtained
`by means of ACK packets. Then, the video sending rate is
`effectively adjusted the target rate ri by choosing an appro-
`priate quantizer scale [8].
`
`2.2 Implementation of MPEG-TFRCP
`To investigate the applicability of the MPEG-TFRCP to the
`actual system, we built the small-scale 10Base-T network as
`illustrated in Fig. 2. The network consists of four personal
`computers, two HUBs and one PC router. Two PCs com-
`municate with each other on TCP connection. The others
`employ UDP or MPEG-TFRCP. Our MPEG-TFRCP sender
`transmits the video data using RTP (Real-time Transport Pro-
`tocol) on the UDP protocol stack and utilizes the RTCP (Real-
`Time Control Protocol) mechanism to obtain the feedback
`information (See Fig. 3). The sender regularly emits a RTCP
`Sender-Report packet to the receiver every 5 pictures. On
`receiving the RTCP packet, the receiver sends back a RTCP
`Receiver-Report packet which contains the number of pack-
`ets it has received since the previous RTCP packet. Then
`the sender can derive the packet loss probability from the
`
`Page 2 of 4
`
`

`

`MPEG-TFRCP
`TCP
`
`0
`
`10
`
`30
`20
`GoP times
`
`40
`
`50
`
`60
`
`Figure 7: Rate variation (Improved)
`
`packet loss probability
`
`1
`
`0.1
`
`loss
`target rate
`video rate
`
`10
`
`02468
`
`0123456789
`
`rate [Mbps]
`
`rate [Mbps]
`
`0
`
`10
`
`40
`
`50
`
`60
`
`0.01
`
`30
`20
`GoP times
`Figure 8: Packet loss probability variation (Improved)
`
`evaluation is 2.6 and higher than Fig. 5 (1.8). Further, the
`video quality is still higher than that of UDP case (2.4).
`We also evaluate the other combination of rate control
`strategies, i.e. Eqs. (1) and (4), or Eqs. (3) and (2). Although
`we do not show those results due to the space limitation, the
`algorithm with Eqs. (3) and (4) provides the most preferable
`control among them.
`
`Investigation into appropriate control interval
`3.2
`As we mentioned in Section 2, our MPEG-TFRCP estimates
`the TCP throughput from the network condition conjectured
`by the obtained feedback information, and then regulates the
`data sending rate at the periodic interval as long as 32×RTT
`(we call this strategy as 32-RTT). The duration of each con-
`trol interval must be carefully determined to attain the ef-
`fective rate control. When the control interval is too short,
`the sending rate changes greatly owing to the frequent rate
`control, then the perceived video quality becomes unstable.
`Conversely, infrequent control leads to the unfairness be-
`cause video applications cannot follow changes of network
`condition.
`In Figs. 9 and 10, employing the rate control mecha-
`nism in Section 3.1, we depicts variations of the averaged
`rate for a few settings of control interval such as 16×RTT
`rounded by GoP-time and 64×RTT rounded by GoP-time,
`respectively.
`In Fig. 9, we depict experimental results of MPEG-
`TFRCP with a 16-RTT interval. When the control interval is
`
`packet loss probability
`
`1
`
`0.1
`
`0.01
`
`14
`
`12
`
`10
`
`loss
`target rate
`video rate
`
`0
`
`10
`
`40
`
`50
`
`60
`
`02468
`
`rate [Mbps]
`
`30
`20
`GoP times
`Figure 6: Packet loss probability variation (MPEG-TFRCP)
`
`stops video transmission as shown in Fig. 5.
`The MPEG-TFRCP sender doubles its data sending
`rate during a loss-free period. As it encounters packet losses,
`it suddenly shrinks the sending rate because the loss proba-
`bility is high due to the network congestion caused by itself.
`Although not shown in figures, the perceived video quality
`considerably fluctuates as the video rate changes. Especially
`when congestion occurs, decreased target rate becomes far
`below the minimum video rate, and no picture can be dis-
`played at the receiver. To solve the drastic rate variation, we
`improve our MPEG-TFRCP in the next section. The con-
`trol intervals are not the same during the session as shown in
`Fig. 6, because the observed RTT fluctuate in an actual sys-
`tem environment. To smooth the RTT variation, we should
`employ some filtering algorithm.
`
`Improving MPEG-TFRCP
`3
`In this section, we consider several methods to improve the
`MPEG-TFRCP mechanism for achieving the higher friend-
`liness and the stable video quality.
`
`3.1 Investigation into rate control algorithm
`The previous work [7] proposes a new rate increase algo-
`rithm, Eq. (4), which imitates the window-based flow control
`of TCP, instead of the rate determination algorithm Eq. (1).
`By the new algorithm, the additive rate increase as in TCP
`can be performed in MPEG-TFRCP. We also replace Eq. (2)
`with the TCP rate decrease algorithm, Eq. (3). Although
`Eq. (1) assumes that statistics are derived from the long-term
`observation and the network condition is relatively steady,
`those are not true on the actual system and it leads to the
`unexpected result (Fig. 5).
`
`
`
`
`ri =
`
`ri−1
`2 ,
`ri−1 + M T U × Ii−1
`
`RT T 2
`
`,
`
`if p > 0
`
`if p = 0
`
`(3)
`
`(4)
`
`The results are depicted in Figs. 7 and 8. As shown
`in those figures, the rate variation becomes relatively small,
`while our MPEG-TFRCP connection can receive the almost
`same throughput as a TCP connection. The subjective video
`quality of Fig. 7 measured by MOS (Mean Opinion Score)
`
`Page 3 of 4
`
`

`

`MPEG-TFRCP
`TCP
`
`0
`
`10
`
`30
`20
`GoP times
`Figure 10: Rate variation (64-RTT)
`
`40
`
`50
`
`60
`
`10
`
`02468
`
`rate [Mbps]
`
`60
`
`MPEG-TFRCP
`TCP
`
`0
`
`10
`
`30
`20
`GoP times
`Figure 9: Rate variation (16-RTT)
`
`40
`
`50
`
`10
`
`02468
`
`rate [Mbps]
`
`shorter in comparison to 32-RTT (Figs. 7 and 8), the sending
`rate extremely changes. This is because the sender cannot
`obtain enough feedback information to accurately estimate
`the network condition.
`In the case of a longer control interval 64-RTT (Fig. 10),
`the rate variation becomes relatively stable. However, the
`sender keeps to send the video data at the higher rate in con-
`trast with TCP connection. The unfair condition tends to
`be hold longer than the control with shorter interval. The
`influence becomes heavier in the real system because RTT
`changes during the session and could suddenly increase as
`shown in Fig. 6.
`Thus, we conclude that the 32-RTT control interval is
`preferable in our system in order that our TFRCP connection
`receives the almost same throughput as a TCP connection
`and the perceived video quality becomes high and stable.
`
`4 Conclusion
`In this paper, we have shown, through real system experi-
`ments, that it is appropriate for real-time video application
`to use a rate control which imitates the TCP window-based
`flow control. We can achieve stable video transfer when
`video rate is regulated at a control interval of 32 times as
`long as RTT. We have found that our MPEG-TFRCP attained
`high-quality and stable video transfer fairly with TCP.
`However, there still remain several research works. First,
`in this paper we have experimented with the small-scale net-
`work. We must consider the large-scale network in which
`a large number of sessions co-exists. Second, though our
`system employ RTCP to obtain the feedback information re-
`quired for the estimation of network condition, those RTCP
`packets may be lost when the network is heavily loaded. In
`that case, our MPEG-TFRCP cannot send video data at the
`proper rate because precision of the network condition es-
`timation degrades. We should adopt some error recovery
`mechanism to avoid the inaccurate estimation. We are cur-
`rently investigating those issues and we will achieve further
`improved MPEG-TFRCP in the near future.
`
`Acknowledgements
`This work was partly supported by Research for the Future
`Program of Japan Society for the Promotion of Science un-
`
`der the Project “Integrated Network Architecture for Ad-
`vanced Multimedia Application Systems”, a Grant-in-Aid
`for Scientific Research (A) 11305030 and a Grant-in-Aid
`for Encouragement of Young Scientists 12750322 from The
`Ministry of Education, Science, Sports and Culture of Japan,
`Special Coordination Funds for promoting Science and Tech-
`nology of the Science and Technology Agency of the Japanese
`Government, and Telecommunication Advancement Organi-
`zation of Japan under the Project “Global Experimental Net-
`works for Information Society Project.”
`
`References
`[1] M. Mathis, J. Semke, J. Mahdavi, and T. Ott, “The
`macroscopic behavior of the TCP congestion avoidance
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`Review, vol. 27, pp. 67–82, July 1997.
`[2] J.-C. Bolot and T. Turletti, “Experience with control
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`SIGCOMM Computer Communication Review, vol. 28,
`pp. 4–15, January 1998.
`[3] J. Padhye, V. Firoiu, D. Towsley, and J. Kurose, “Mod-
`eling TCP throughput: A simple model and its empir-
`ical validation,” Proceedings of ACM SIGCOMM’98,
`vol. 28, pp. 303–314, October 1998.
`[4] J. Padhye, J. Kurose, D. Towsley, and R. Koodli,
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`[6] N. Wakamiya, M. Murata, and H. Miyahara, “On
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`application-level QoS,” to be presented at IEEE ICME
`2000, July 2000.
`[7] N. Wakamiya, M. Murata, and H. Miyahara, “On TCP-
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`2000.
`[8] K. Fukuda, N. Wakamiya, M. Murata, and H. Miyahara,
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`
`Page 4 of 4
`
`

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