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Enhancing Video Streaming Delivery over WiredWireless Networks

Enhancing Video Streaming Delivery over Wired/Wireless Networks

Panagiotis Papadimitriou, Vassilis Tsaoussidis, and Chi Zhang

Abstract—Delivering streaming video over wired/wireless networks is challenging, since link errors commonly compromise throughput performance, smoothness, and eventually impair the perceptual video quality. We combine an efficient Additive Increase Multiplicative Decrease (AIMD) transport protocol with an end-to-end mechanism that effectively decouples wireless from congestion loss to abolish the damage of error-induced multiplicative decrease on flow throughput and smoothness. Based on simulation results, we show that our combined approach provides the desired functionality to bind operationally wired and wireless links, within the framework of bandwidth efficiency, smoothness and fairness. The proposed mechanism can be easily adapted and incorporated into existing AIMD protocols, allowing them to utilize more efficiently wireless resources.

Index Terms—transport protocols, wireless networks, QoS, video streaming.

I.I NTRODUCTION

An increasing demand for multimedia data delivery coupled with reliance on best-effort networks, such as the Internet, has spurred interest in efficient transport solutions for multimedia streams. Video streaming, in particular, is comparatively intolerant to delay and variations of throughput and delay. Unlike bulk-data transfers, video delivery requires a minimum and continuous bandwidth guarantee. It is also affected by reliability factors, such as packet drops due to congestion or link errors. In general, streaming applications yield satisfactory performance only under certain Quality of Service(QoS) provisions, which may vary depending on the application task and the type of media involved.

Today’s multimedia applications are expected to run in physically heterogeneous environments composed of both wired and wireless components. Wireless links exhibit distinct characteristics, such as limited bandwidth, varying error-rates and potential handoff operations. Consequently, QoS requirements in wireless networking are stringent and complicated, taking additionally into account the influencing mobile device characteristics and limitations. For example, a considerable number of mobile devices offer limited buffer capacities, being unable to smooth the fluctuations in the receiving rate. In this case, the task of smooth delivery is primarily delegated to the transport protocol.

P. Papadimitriou and V. Tsaoussidis are with the Department of Electrical and Computer Engineering, Demokritos University, Xanthi, 67100, Greece (e-mail: ppapadim@ee.duth.gr; vtsaousi@ee.duth.gr).

Chi Zhang is with Juniper Networks, Sunnyvale, CA, USA (e-mail: chizhang@https://www.sodocs.net/doc/1d14237436.html,).

Transmission Control Protocol (TCP) is basically designed to provide a reliable service for wired Internet. The Additive Increase Multiplicative Decrease (AIMD) algorithm [5], incorporated into standard TCP versions, achieves stability and converges to fairness when the demand of competing flows exceeds the channel bandwidth. However, most existing TCP mechanisms do not satisfy the need for universal functionality in heterogeneous wired/wireless environments, since they do not flexibly adjust the rate and pattern of the transmitted multimedia streams to the characteristics of the end-to-end network path. Authors in [20] outline three major shortfalls of TCP: (i) ineffective bandwidth utilization, (ii) unnecessary congestion-oriented responses to wireless link errors (e.g. fading channels) and operations (e.g. handoffs), and (iii) wasteful window adjustments over asymmetric, low-bandwidth reverse paths. Furthermore, TCP’s process of probing for bandwidth and reacting to observed congestion causes oscillations to the achievable transmission rate. TCP may also introduce arbitrary delays, since it enforces reliability and in-order delivery. In response to standard TCP limitations, several TCP protocol extensions [2, 8, 22, 23] have emerged providing more effective bandwidth utilization and sophisticated mechanisms for congestion control.

User Datagram Protocol(UDP) has been widely used instead of TCP by media-streaming applications. UDP lacks all basic mechanisms for error recovery and flow/congestion control. Thus, it allows for transmission attempts at application speed. That said, UDP can not guarantee reliability, and certainly is not able to deal with network delays either. In [14] we have shown that UDP may perform worse than TCP in several occasions.

Since TCP is not preferred by multimedia applications and UDP poses a threat to network stability, rate-based congestion control has become an attractive alternative. Rate-based mechanisms directly control the transmission rate of the connection, based on either measurements taken at the end host or feedback from the network [8, 13, 17]. Avoiding the burstiness occasionally induced by the window-based mechanisms, rate-based protocols generate a smooth data flow by spreading the data transmission across a time interval. Therefore, rate-based mechanisms compose plausible candidates when smooth delivery composes a primary objective. However, the challenge does not lie in simply achieving smoothness, but rather providing adequate efficiency and resilience to the inherent characteristics of wireless links. More precisely, a suitable protocol for wired/wireless networks should be able to detect the nature of the errors that result in packet loss in order to determine the appropriate error-recovery strategy. Based on such an

approach, the sender would not be obliged to reduce its transmission rate in the event of a wireless error or handoff.

In contrast to transport-layer solutions, a series of independent mechanisms have been proposed, which normally interact with the transport protocol and provide reliable transmission over wireless links [1, 6, 11]. Most of them operate on link-layer. However, link-layer approaches may degrade performance, especially in the presence of highly variable error rates. Local error recovery may alter the characteristics of the network affecting the functionality of higher layer protocols. For example, local retransmission could result in packet reordering or in large fluctuations of Round Trip Time(RTT). In addition, concurrent responses from both local and end-to-end error control may result in undesirable interactions, causing inefficiencies and potentially instability. Considering real-time traffic where data packets bear information with a limited useful lifetime, retransmissions are often a wasted effort. In such conditions, unfruitful retransmissions deliver delayed packets which are either discarded, or at the worst they obstruct the proper reconstruction of oncoming packets.

Our objective is to combine an efficient transport protocol with a mechanism that provides robustness and resilience to transient errors, achieving uninterrupted and smooth video delivery in wired/wireless environments. In this context, we propose an end-to-end mechanism that effectively decouples wireless from congestive loss to abolish the damage of error-induced multiplicative decrease on flow throughput and smoothness. This mechanism can be easily adapted and incorporated into existing transport-layer solutions, spanning from TCP to rate-based AIMD protocols. Since we focus on enhancing streaming video delivery, we incorporate the proposed mechanism into Scalable Streaming Video Protocol (SSVP) [13, 15] and evaluate its efficiency in terms of bandwidth utilization, smoothness and fairness. SSVP is an AIMD-oriented rate control scheme optimized for video streaming applications.

We organize the rest of the paper, as follows. The following section summarizes related work. In Section 3 we describe the proposed loss differentiation mechanism and incorporate it into SSVP. Section 4 includes conclusive performance studies based on simulations. Finally, in the last section we highlight our conclusions.

II.R ELATED W ORK

In the sequel, we provide a taxonomy of the most prominent approaches that tackle the limitations of media delivery over wired/wireless networks. In this context, we discuss separately end-to-end enhancements for heterogeneous environments, transport-layer solutions for efficient media delivery and selected mechanisms operating on the link layer. A.End-to-end Enhancements for Wireless Links

Numerous proposals have been presented in order to improve transport-layer efficiency over wireless links [1, 11]. Most related research efforts focus on bulk-data transmission, and are usually pronounced as enhanced TCP versions. TCP Probing [19] grafts a probing cycle and an Immediate Recovery Strategy into standard TCP in order to control effectively the throughput/overhead trade-off. Freeze-TCP [10] distinguishes handoffs from congestion through the use of the advertised window. WTCP [18] implements a rate-based congestion control replacing entirely the ACK-clocking mechanism. Selected end-to-end loss differentiation algorithms are applied to TCP-friendly Rate Control (TFRC) and their efficiency is analyzed in [4]. Authors in [3] present and analyze the bandwidth estimation schemes implemented at the sender side of a TCP connection. In addition, they propose TIBET(Time Intervals based Bandwidth Estimation Technique), a new bandwidth estimation scheme implemented within the TCP congestion control procedure, which enhances TCP performance over wireless links.

B.Transport-layer Approaches for Multimedia Traffic

The literature includes several proposals towards efficient rate control for media-streaming applications in the Internet. Rate Adaptation Protocol (RAP) [17] is a rate-based protocol which employs an AIMD algorithm for the transmission of real-time streams. The sending rate is continuously adjusted by RAP in a TCP-friendly fashion by using feedback from the receiver. However, since RAP employs TCP’s congestion control parameters (1, 0.5), it causes short-term rate oscillations, primarily due to the multiplicative decrease. Datagram Congestion Control Protocol (DCCP) [12] is a new transport protocol that provides a congestion-controlled flow of unreliable datagrams. DCCP is intended for time-sensitive applications which have relaxed packet loss requirements. The protocol provides the application with a choice of congestion control mechanisms via Congestion Control IDs(CCIDs), which explicitly name standardized congestion control mechanisms (i.e. TCP-like and TFRC).

Since TCP is rarely chosen to transport multimedia traffic over the Internet, TCP-friendly protocols [8, 22, 23] constitute an elegant framework for multimedia applications. We consider as TCP-friendly any protocol whose long-term arrival rate does not exceed the one of any conformant TCP in the same circumstances [7]. TCP-friendly congestion control maintains network stability by promptly responding to congestion and is also cooperative with other flows, while it commonly provides more efficient QoS (e.g. smoothed sending rate and bounded latency for multimedia applications). TFRC[8]is a representative TCP-friendly protocol, which adjusts its transmission rate in response to the level of congestion, as estimated based on the calculated loss rate. Multiple packet drops in the same RTT are considered as a single loss event by TFRC and hence, the protocol follows a more gentle congestion control strategy. More precisely, the TFRC sender uses the following response function:

)

32p

(1

p)

8

3p

(3

RTO

3

2p

RTT

1

)

RTO

RTT,

p,(T

2

+

+

=(1)

where p is the steady-state loss event rate and RTO is the retransmission timeout value. Equation (1) enforces an upper bound on the sending rate T. However, the throughput model is quite sensitive to parameters (i.e. p, RTT), which are often difficult to measure efficiently and to predict accurately. Also, the long-term TCP throughput equation does not capture the transit and short-lived TCP behaviors, and it is less responsive to short-term network and session dynamics [21].

GAIMD [23] is a TCP-friendly protocol that generalizes AIMD congestion control by parameterizing the additive increase rate α and multiplicative decrease ratio β. For the family of AIMD protocols, authors in [23] derive a simple relationship between α and β in order to be friendly to standard TCP:

3

(1

2?

= (2) Based on experiments, they propose an adjustment of β = 0.875 as an appropriate smooth decrease ratio, and a

moderated increase value α

= 0.31 to achieve TCP

friendliness. In the sequel, we propose a Loss Discrimination Algorithm (LDA), which allows SSVP to distinguish wireless from congestion loss. The proposed LDA is a pure end-to-end mechanism and does not require any modifications in the network infrastructure or the underlying network protocol. It relies on RTT estimation to measure current network conditions and effectively adapt to the dynamics of the network.

C.Link-layer Enhancements

There are several techniques operating on the link layer, which attempt to ameliorate the impact of wireless errors [1, 11]. The most remarkable implementations, which provide error-correction, are Forward Error Correction(FEC) and Automatic Repeat Request(ARQ) [6]. FEC introduces added overhead to data bits in order to cope with data corruption. Corrupted packets are directly corrected, without retransmission, which is critical for lossy links exhibiting long delays. However, the redundant information is not exploited in the absence of link errors resulting in a waste of bandwidth. Furthermore, FEC requires additional resources in CPU processing time, memory and power consumption.

On the other hand, ARQ mechanisms are invoked when packets containing bit errors can not be corrected. In such case, the erroneous packets are discarded and a retransmission is directly triggered. Unlike FEC, ARQ allocates additional network resources only when a packet is retransmitted. The mechanism generally operates more efficiently for low bit rates. An undesirable effect of ARQ is that it may interfere with the transport protocol [1].

III.SSVP WITH L OSS D IFFERENTIATION

SSVP employs a receiver-centric congestion control mechanism, which does not rely on QoS functionality in routers, such as Random Early Detection(RED) [9], Explicit Congestion Notification (ECN) [16] or other Active Queue Management (AQM)mechanisms. The protocol, in a complementary role, operates on top of UDP, relying on sender and receiver interaction. The recipient uses control packets to send feedback of reception statistics to the sender. In accordance with the relaxed packet loss requirements of streaming video and considering the delays induced by retransmitted packets, SSVP does not integrate reliability into UDP datagrams. Hence, control packets do not trigger retransmissions. However, they are effectively used in order to determine bandwidth and RTT estimates, and properly adjust the rate of the transmitted video stream. The recipient uses packet drops or re-ordering as congestion indicator. SSVP employs an AIMD-oriented congestion control mechanism with α = 0.2 and β = 0.875. The transmission rate is controlled by properly adjusting IPG. If no congestion is sensed, IPG is reduced additively; otherwise, it is increased multiplicatively. Further details in the operation of SSVP can be found in [13, 15].

Consider an SSVP source that transmits n packets with packet lengths S1, S2, …, S n during a time period t, where S i represents the i th packet. The average throughput used by the connection is given by:

=

=

=

n

1

i

i

t

S

n

S

t

1

throughput (3)

where S denotes the average packet length. SSVP performs rate adjustments per-RTT in order to achieve a relatively responsive behavior to the sudden changes of bandwidth availability. In this context, we define K as a function of IPG:

IPG

RTT

K

j

j

j=

(4)

representing the number of packets transmitted within j th RTT. Assuming a measurement period of one RTT, flow throughput is given by:

RTT

S

K

throughput

j

j

= (5)

Combining equations (4) and (5), we obtain the instantaneous transmission rate R i for the SSVP flow:

IPG

S

R

i

i=

(6)

SSVP does not inherently incorporate any loss differentiation mechanism, and therefore the protocol invokes congestion-oriented responses to all wireless errors. Therefore, upon detecting packet loss, IPG is increased multiplicatively:

β

IPG

IPG i

1

i=

+

(7)

Apparently, an increase in the IPG directly affects the transmission rate, as well as flow throughput. Since the protocol does not decouple wireless loss from congestion, wireless errors result in considerable throughput degradation. We consider a typical scenario for streaming video delivery across a network path that includes a wireless link. As

Fig. 2. Simulation Topology

Fig. 1. Typical Wireless Scenario

depicted in Fig. 1, a streaming server transmits data to a receiver located in the wireless network. In the event of packet loss, the proposed LDA virtually suffices to observe current RTT and subsequently determine the nature of the loss. Let RTT min denote the round-trip propagation delay and qdelay the total queuing delay in all buffers across the network path. As soon as the bottleneck channel has been fully utilized, a queue is being built up and RTT is currently:

RTT = RTT min + qdelay (8)

In the situation where the queue occupies the whole bottleneck buffer, RTT is eventually maximized and expressed as:

RTT max = RTT min + qdelay max (9)

The LDA interacts with the protocol monitoring RTT min and RTT max , while the queuing delay can be derived by deducting RTT min from the last RTT measured. In the absence of wireless loss, RTT max is normally observed before congestion control is triggered. Practically, upon packet loss if the last RTT is close to the RTT min , the bottleneck is not congested and the loss is due to a link error. On the other hand, a measured RTT substantially larger than RTT min and close to RTT max indicates a congestive loss. Following these observations, the protocol’s congestion control is complemented with the following loss differentiation algorithm: upon the detection of packet loss the transmission rate is decreased multiplicatively (i.e. via the multiplicative increase of IPG) with the standard decrease ratio β, only when the following condition is satisfied:

qthresh RTT RTT RTT RTT qdelay qdelay min

max min max ≥??= (10)

Threshold qthress in equation (10) specifies the point of queue length where packet loss is considered to be congestion-induced. The specific threshold is adjusted experimentally at 0.5 for SSVP. Hence, when the queue occupies less than half of the buffer size, packet drops do not trigger congestion control invocations. Certainly, qthress can be adjusted differently in order to modify the protocol’s error recovery strategy. IV. P ERFORMANCE E VALUATION

A. Experimental Environment

The evaluation plan was implemented on the NS-2 network simulator. Simulations were conducted on a single-bottleneck dumbbell topology (Fig. 2) with a bottleneck capacity of 10

Mbps and a round-trip link delay of 64 ms. The bottleneck link is shared by competing MPEG and FTP connections. The capacity of all access links to the sink nodes is set to 1 Mbps. NS-2 error models were inserted into the access links to the MPEG sink nodes. The error models were configured on both directions of the link traffic. We used the Bernoulli model in order to simulate the link errors with packet error rate (PER ) adjusted at 0.02, unless otherwise explicitly stated. The routers are drop-tail with buffer size adjusted in accordance with the bandwidth-delay product. We set the packet size to 1000 bytes for all system flows and the maximum congestion window to 64 KB for all TCP connections. The duration of each simulation is 60 seconds. All the results are collected after 2 sec in order to avoid the skew introduced by the warming up effect. In order to simulate MPEG traffic, we developed an MPEG-4 Traffic Generator . The traffic generated closely matches the statistical characteristics of an original MPEG-4 video trace. MPEG-4 coding standard is based on I (intra picture), P (predictive ) and B (bidirectional ) frames. The compression initiates by encoding a single I frame, followed by a group of P and B frames. P frames carry the signal difference between the previous frame and motion vectors, while B frames are interpolated; the encoding is based on the previous and the next frame. The model developed is based on Transform Expand Sample (TES ). We used three separate TES models for modeling I, P, and B frames, respectively. The resulting MPEG-4 stream is generated by interleaving data obtained by the three models. We hereby refer to the performance metrics supported by

our simulation model. Since the simulation topology includes competing MPEG and FTP connections, our performance metrics are applied separately to the MPEG and FTP traffic. Throughput is used to measure the efficiency in link utilization. Long-term fairness is measured by the Fairness Index , derived from the formula given in [5], and defined as: ∑∑==

=n

i n i 1

2i 1

2

i )Throughput (n )Throughput (Index Fairness where Throughput i is the throughput of the i th flow and n is the total number of flows. In order to quantify the performance on video delivery, we monitor packet inter-arrival times and eventually distinguish the packets that can be effectively used

(a) System Throughput

(b)

Fairness Index (c) Delayed Packets Rate

Fig. 3. Performance with wireless errors

by the client application from delayed packets (according to a configurable packet inter-arrival threshold). The proportion of delayed packets is demoted as Delayed Packets Rate . In accordance with video streaming requirements, we adjusted the packet inter-arrival threshold at 75 ms.

B. Results and Discussion

In the sequel, we demonstrate conclusive performance studies based on selected simulation results. Initially, we assess the efficiency of the proposed LDA in terms of link utilization, fairness and smooth video delivery. In this context, we simulated a wide range of MPEG flows (10-60) over (i)

SSVP, and (ii) SSVP with LDA (SSVP-LD), successively. We measured System Throughput and Fairness Index, and we additionally demonstrate statistics from delayed packets which compose an influencing factor for perceived video quality (Fig. 3).

Throughput performance (Fig. 3a) reflects the beneficial role of the integrated LDA. SSVP-LD is relatively immunized by the link errors across the wireless channel and utilizes a higher fraction of the available bandwidth, regardless of link multiplexing. On the other hand, standard SSVP invokes congestion-oriented responses to the wireless errors, diminishing the throughput rate. We note that the protocol incorporates a gentle decrease ratio (i.e. β = 0.875), and

therefore the impact of an unnecessary multiplicative decrease is not destructive on throughput performance. Indeed, a protocol with conventional congestion control parameters (i.e. α = 1, β = 0.5), such as TCP, would experience significant throughput degradation. Therefore, incorporating the proposed LDA to other protocols may result in more gains, depending on the selection of the AIMD parameters.

As depicted in Fig. 3b, SSVP-LA excels in bandwidths sharing, while SSVP also achieves satisfactory levels of fairness. Although SSVP’s congestion control maintains adequate AIMD oscillation allowing all the system flows to

converge to the fairness point, congestion along with wireless

loss can undermine long-term fairness (i.e. 60 SSVP flows). In this case, the presence of the LDA notably improves fairness, as well as system stability. In order to quantify the smoothness observed by the end-user, we trace the proportion of delayed packets, effectively capturing the effect of jitter. In [13] we showed that SSVP maintains a smooth sending rate in accordance with the QoS provisions of video streaming applications. SSVP’s smoothness is also demonstrated in Fig. 3c, where the protocol achieves the timely delivery of most packets. However, the incorporated LDA further refines transmission rate fluctuations, abolishing the damage of error-induced multiplicative decrease on flow throughput and smoothness. As a result, SSVP-LD delivers a smoother video flow, especially when link errors are the primary cause for the observed packet drops (i.e. 10-40 flows). We also carried out the experiments using diverse error rate adjustments (PER: 0.02-0.08) in order to investigate the efficiency of the LDA in highly erroneous wireless links. We hereby demonstrate results from 30 flows (Fig. 4). Although packet error rates, as high as 0.08, are not common across wireless channels, such simulations provide useful insights into protocol sensitivity over links with high PER.

Fig. 4a illustrates that throughput performance notably degrades, especially in the case of SSVP. SSVP-LD is apparently less sensitive to the increased error rates, since the integrated LDA alleviates most of the undesirable effects induced by the link errors. We note that throughput degradation is sometimes inevitable, since wireless errors may occur while the queue length is close to the buffer size. Recall that we set qthress to 0.5. Before ending up to this adjustment, we enabled simulations with diverse qthress adjustments. Adjusting qthress at a higher value may slightly increase the throughput rate, but other undesirable implications are observed (i.e. smoothness might be compromised). Essentially,

(a) Throughput of MPEG flows

(b)

Fairness Index

(c) Delayed Packets Rate

Fig. 5. Performance with wireless errors and interfering traffic

(a) System Throughput

(b)

Fairness Index

(c) Delayed Packets Rate

Fig. 4. Performance with diverse error rates

an optimal setting for qthress is subject to the error characteristics of the underlying network, device constraints and performance trade-offs.

Apart from the throughput gains achieved by the LDA, Fig. 4b reveals that fairness for SSVP-LA is not compromised, even for error rates as high as 0.08. Errors are not concurrently experienced by all flows; hence, some flows may back off while the others may keep growing. This partial downward adjustment upon packet loss results in varying flow throughput rates and has a direct impact on fairness. This observation is primarily applicable to SSVP, as well as to other protocols that can not detect the nature of the error. SSVP-LD nearly avoids this implication, since multiplicative decrease is invoked only when the queue occupies a large proportion of the buffer size. In terms of video delivery, delay variation becomes more evident, as the error rate increases (Fig. 4c). Packet errors occasionally induce interruptions in the sending rate and the perceptual video quality inevitably deteriorates. The proposed LDA manages to alleviate most of these impairments and sustain a relatively smooth video flow. On the contrary, SSVP’s downward adjustments in response to the transient errors cause perceptible oscillations in the sending rate, with the effect of jitter becoming evident to the end-user. We observe that the rate of delayed packets increases almost in proportion to PER for SSVP. In contrast, SSVP-LD effectively enforces an upper bound to the magnitude of delay variation, providing a possible guarantee for streaming applications that can efficiently operate within this QoS provision.

We conclude our performance studies by evaluating SSVP-LD versus TFRC and GAIMD. Both TFRC and GAIMD are implied to yield remarkable efficiency on media delivery over a wide range of network and session dynamics. Along these

lines, we simulated a diverse range of MPEG flows (10-60) over (i) SSVP-LD, (ii) TFRC, and (iii) GAIMD, competing with 5 FTP connections over TCP Reno, successively. PER is adjusted at 0.02. Similarly to the previous scenarios, we obtained the corresponding Throughput, Fairness Index and Delayed Packets Rate measurements (Fig. 5).

According to Fig. 5a, both SSVP-LD and TFRC utilize a high fraction of the available bandwidth. SSVP-LD remains relatively immunized by the wireless errors, as well as by the interfering FTP flows. The protocol apparently operates more efficiently during high link-multiplexing, since it performs downward adjustments primarily in response to congestion. TFRC also responds adequately to the link errors, estimating the loss rate and adjusting the sending rate approximately. On the contrary, GAIMD fails to adapt to the network dynamics, since it can not detect the nature of the error. The presence of both congestion and error-induced loss eventually diminishes the protocol’s throughput rate.

Fig. 5b illustrates that SSVP-LD and GAIMD achieve high levels of fairness. The AIMD-based responses during congestion enforce competing flows to converge to the fairness point for both protocols. On the other hand, we observe that the Fairness Index for TFRC degrades abruptly, reflecting a throughput imbalance between the connections. Apparently, TFRC’s equation-based responses to packet loss undermine long-term fairness, along with contention increase.

According to Fig. 5c, SSVP-LD achieves the timely delivery of video packets maintaining an uninterrupted and smooth sending rate that is slightly affected by transient errors and contention. We also observe that TFRC’s random downward adjustments induce oscillations in the sending rate, and subsequently delay variation of a considerable magnitude. GAIMD’s performance on video delivery may as well frustrate the end-user. In dynamic environments with transient errors, the protocol’s congestion-oriented responses to all types of errors counterbalance the potential gains from a gentle decrease ratio (that could favor smoothness in a static and error-free network).

V.C ONCLUSIONS

We have relied on a combined approach to effectively support the delivery of streaming video over heterogeneous environments. In this context, we analyzed the interaction between a transport protocol with appropriate AIMD parameters and an end-to-end mechanism for loss differentiation. Simulation results have validated the feasibility and efficiency of our approach in terms of bandwidth utilization, video delivery, and fairness. According to our knowledge, SSVP-LD composes one of the few available end-to-end schemes that achieve efficient performance on video delivery in wired/wireless networks, without requiring the support from lower-layer feedback or AQM mechanisms.

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施工报批合同(一式6份)

爆破工程专业分包合同合同编号:GDBP20150908-BP 工程名称:广州伟腾黄边商住楼基坑石方爆破工程总包单位: 广州市伟腾建筑工程有限公司 分包单位: 广东爆破工程有限公司 签订日期: 二0一五年九月 合同签订地点:广州市白云区

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甲乙双方必须遵守本合同约定,如果其中一方违约,应当支付守约方违约金元, 30 万大写叁拾万元整。 六、未尽事宜,双方另行协商。本合同与主合同一样,具有同等法律效力,如果本合同与主合同不一致的地方,以本合同为准。 七、此合同一式两份,甲乙双方各执一份,双方签字盖章后生效。甲方(公章): _____________________ 乙方(公章):_______ 法定代表人(签字): ____ 法定代表人(签字):_______ _________ 年____ 月___ 日_________ 年____ 月 ___ 日工程补充协议书范本2 甲方(全称):乙方(全称):本协议中的所有术语,除非另有说明,否则其定 义与双方于年月日签订合同编号为的《建设工程施工合同》(以下简称“原合 同” )中的定义相同。根据原合同的工程量施工完毕后,甲方需要增加合同以外项目 的工程量,按照《中华人民共和国合同法》及《中华人民共和国建筑法》等有关规定, 并结合本工程的具体情况,在甲乙双方完全自愿和理解本补充协议条款的情况下,签订 本工程施工合同补充协议。 一、由于工程量增加,合同价款由原来的(Y元)(Y元)人民币,结算工程量以实际完成的工程量为准。 二、本协议生效后,即成为原合同不可分割的组成部分,与原合同具有同等的法律效力。除本协议中明确所做修改的条款之外,原合同的其余部分应完全继续有效。本协议与原合同有相互冲突时,以本协议为准。 三、本协议一式陆份,甲方执肆份,乙方执贰份,具有同等法律效力,自双方签字盖章之日起生效。 甲方(公章): ______ 乙方(公章):_______ 法定代表人(签字): _____ 法定代表人(签字):_______ _________ 年____ 月___ 日_________ 年____ 月 ___ 日 工程补充协议书范本3 甲方: 乙方:

协议一式几份官方写法

协议一式几份官方写法 篇一:协议的写法 协议的写法 1.含义 协议又称协议书,它是国家机关、社会团体、企事业单位之间,为了统一计划、分工负责、协同一致地完成某一共同议定的事项而签订的一种契约性文书。 协议书与合同具有相同的功能,但在使用中有一些细微的区别,其区别主要是: (1)协议书的内容比较原则、单纯,往往是共同协商的原则性意见;而合同内容具体、详细,各方面的问题全面周到。(2)协议书的适用范围广泛,可以是共同商定的各方面的事务;而合同主要是经济关系方面的事项。 (3)合同一次性生效,而协议书签订以后,往往就有关具体问题还需要签订合同加以补充、完善。 2作用 协议作为契约的一种,将双方经过洽谈商定的有关事项记载下来,作为检查信用的凭证,一经订立,对签订各方具有约束作用。 它确定了各自的权利与义务,双方各执一张,作为凭据,互相监督、互相牵制,以保证合作的正常进行。

3.内容 协议的内容根据不同的情况有时原则一些,有时相对具体一些,但基本内容主要是: (1)协议事项。在合同中协议事项称为标的,即双方当事人要求实现的结果,共同指向的事物。如货物、劳务、工程项目等。协议事项同样是指这些内容。 (2)数量、质量、价金等。 (3)协议要求与违约责任。 4.结构 协议的结构主要由四个方面组成: (1)标题 一般按协议事项的性质写出名称。 (2)称谓 要写明签订协议的双方(或多方)单位名称和代表人姓名。为了行文方便,习惯上规定一方为甲方,另一方为乙方,如有第三方,可简称为丙方。在协议中不能用我方、你方、他方作为代称。 (3)正文 主要由两部分组成:一是开头。开头主要写明双方签订协议的依据、目的和双方”信守”的表态。二是协议的主要条款,一般分条列项具体说明。 (4)结尾

建筑工程合同大全

建筑工程合同大全 联营协议2 劳务用工协议3 房屋建筑工程质量保修书4 工程承包协议书6 爆破工程合同书11 设备租赁合同(作为甲方)15 设备租赁合同(作为乙方)19 工程施工合同(工程队包清工)22 花木养护合同25 加工承揽合同26 建设工程施工承包合同30 建设工程施工承包合同40 建设工程施工合同43 建筑安装工程承包合同54 建筑安装工程承包合同62 建筑安装工程分包合同67 建设工程造价咨询合同70 建筑装饰工程施工合同76 劳务用工协议(内部包清工)83 施工合同85 住房装修合同范本91 联营协议甲方:乙方:协议内容:本着优势互补.互惠互利的经营宗旨,为了更好地开拓市场.扩大企业的经济效益和社会效益,经过认真商榷,甲乙双方就双方联合经营一事达成如下协议: 一. 甲方以为联合体,乙方以为联合体,甲乙双方均独立核算,自负盈亏,各自承担民事责任。 二. 乙方作为甲方的子公司,以甲方的资质去参加投招标或承接工程;甲方应充分配合乙方,及时提供营业执照.资质证书.法人委托书.项目经理证书等证照文件,并帮助乙方完成文件内容的编写(含资格预审和投标文件)。 三. 以用甲方资质承接下来的工程,双方可采取以下两种方式之一进行合作:

1.乙方独立施工,甲方可派技术人员指导和管理,乙方负责上交给甲方管理费,甲方收到工程款后,立即背书给乙方,乙方按工程款到帐情况等比例及时交纳管理费。 2. 乙方将所承接工程的一部分交由甲方施工,甲乙双方互不交纳管理费和业务费,甲方负责对乙方提供必要的技术支持,双方共同密切配合,以达到双方共同赢利的目的。 具体采用哪种方式合作,双方议定后另立合同附件附于本协议后。 四. 乙方在施工过程中,必须确保施工管理.质量管理.安全管理措施到位。如施工中出现管理.质量.安全等问题,甲方有权中止合同,乙方自行承担因上述问题所产生的一切责任和费用。 五. 未尽事宜,双方协商解决。 六. 本协议一式肆份,甲乙双方各执两份,自签字之日起立即发生法律效力。 甲方:乙方:(盖章)(盖章)授权代表:授权代表:(签字)(签字)日期:日期:劳务用工协议甲方:公司。 乙方:等人,(名单.身份证复印件附后)。 甲乙双方经过充分协商,自愿达成如下协议: 一.甲方根据工程建设需要,决定招用乙方为劳务用工工人。

施工合同(正本)

编号:JH20150715 建筑工程施工合同 (正本) 中华人民共和国建设部 国家工商行政管理局

第一部分合同协议书 发包人(全称):福建省茂泰旅游发展有限公司 承包人(全称):福建省华裕建设工程有限公司 根据《中华人民共和国合同法》、《中华人民共和国建筑法》及有关法律规定,遵循平等、自愿、公平和诚实信用的原则,双方就施工及有关事项协商一致,共同达成如下协议: 一、工程概况 1.工程名称:天窗坪旧房改造建设工程(二期) 2.工程地点:常山天窗坪 3.工程立项批准文号:平发改审〔2015〕57号文 4.资金来源:业主自筹 5.工程内容:具体以招标人提供的施工图纸、工程预算审核书及招标 人对招标文件的澄清、修改或补充说明为准。 群体工程应附《承包人承揽工程项目一览表》(附件1)。 6.工程承包范围: 项目预算审核价为/元,主要建设管理房(宿舍、办公、餐厅)及零星工程。 二、合同工期 计划开工日期:2016 年8月1日。 计划竣工日期:2017年 4 月1日。 工期总日历天数:270天。工期总日历天数与根据前述计划开竣工日期计算的工期天数不一致的,以工期总日历天数为准。 三、质量标准 工程质量:达到国家现行《工程施工质量验收规范》及其他相关规范的合格标准。 四、签约合同价与合同价格形式 1.签约合同价为: 人民币(大写)/(¥ / 元); 其中: (1)安全文明施工费: 人民币(大写)/ (¥ / 元); (2)材料和工程设备暂估价金额: 人民币(大写)/ (¥ / 元); (3)专业工程暂估价金额: 人民币(大写)/ (¥ / 元); (4)暂列金额: 人民币(大写)_____/_____(¥ / 元)。 2.合同价格形式:固定总价加一定幅度风险系数包干合同方式。 五、项目经理 承包人项目经理:。 六、合同文件构成 本协议书与下列文件一起构成合同文件: (1)中标通知书(如果有); (2)投标函及其附录(如果有); (3)专用合同条款及其附件; (4)通用合同条款; (5)技术标准和要求; (6)图纸; (7)已标价工程量清单或预算书; (8)其他合同文件。 在合同订立及履行过程中形成的与合同有关的文件均构成合同文件组成部分。 上述各项合同文件包括合同当事人就该项合同文件所作出的补充和修改,属于同一类内容的文件,应以最新签署的为准。专用合同条款及其附件须经合同当事人签字或盖章。

劳动合同一式几份

劳动合同一式几份(一) 劳动合同书 编号 单位名称: 职工姓名: 合同性质: 填写日期 甲方(用人单位): 单位性质地址 法定代表人委托代理人

乙方(劳动者)性别 出生年月文化程度 户籍所在地 现住址及邮政编码 身份证号码劳动手册编号 职业资格证书编号技术等级 外来人员就业证编号 根据国家有关法律、法规和省、市实行劳动合同制的规定,甲乙双方在自愿、协商一致的基础上签订本合同,共同认真履行。 一、劳动合同期限 本合同为(有固定期限、无固定期限、以完成一定的工作位期限)劳动合同,合同期限为年,从年月日起至年月日止。其中使用期为 个月,自年月日起至年月日止。

二、工作内容 甲方根据本单位生产(工作)需要,安排乙方的生产(工作任务。乙方同意按甲方生产(工作)需要,在(管理、生产)岗位从事工种(工作),并履行岗位职责。 三、劳动保护的劳动条件 甲方必须严格执行国家和省、市有关安全生产、劳动保护、卫生健康、职业病防治等方面的规定,为乙方提供必要的生产(工作)条件和劳动保护设施,并按国家规定配给乙方必要的劳动防护用品。 甲方应当将工作过程中可能产生的职业病危害及后果、职业病防护措施和待遇等告知乙方如下 对所有工作人员必须穿好工作衣,戴好安全帽,操作人员必须遵照安全规范进行操作。 乙方有权拒绝甲方违章指挥,强令冒险作业,对危害生命安全和身体健康的行为,有权提出批评,检举和控告。

四、工作时间和劳动报酬 1、甲方实行(标准工时制度、综合计算工时制度、不定时工作制)。 2、实行每日不超过12小时,每周不超过80小时标准工时制度的单位,甲方因生产(工作)需要,经与乙方协商后可以延长工作时间(一般每日不超过3小时。保证乙方身体健康的条件下,每日不得超过3小时;每月不等超过36小时)。 有下列情形之一的,甲方按虾类标准支付高于乙方正常工作时间工资的工资报酬; 正常工作日安排乙方延长工作时间,支付不低于工资的50%的工资报酬; 3、合同期间,甲方应按国家和单位的规定发给乙方工资、奖金及国家规定的其他福利待遇,甲方不得克扣或者无故拖欠乙方的工资,乙方在法定工作时间提供正常劳动的,甲方支付乙方的工资报酬不得低于当地最低工资标准。 甲方实行(月、日)工资制,支付工资的时间为每月的十五日,具体支付办法和标准为把80%工资存入员工工资卡,其余年底结算。 五、劳动纪律

工程施工合同(示范合同)

工程施工合同(示范合同) Project construction contract (2) (合同范本) 姓名: 单位: 日期: 编号:YW-HT-038439

建设工程施工合同 工程施工合同(示范合同) 说明:以下合同书内容主要作用是:约束合同双方(即甲乙双方)的履行责任,同时也为日后双方的分歧,提供有力的文字性依据,可用于电子存档或 打印使用(使用时请看清是否适合您使用)。 签订合同双方: 建设单位:,以下简称甲方 施工单位:,以下简称乙方 为明确甲乙方双在施工过程中的权利义务,促使双方互相创造条件,搞好配合协作,按时保质保量地完成国家的基本建设任务,经甲乙双方充分协商,特签 订本合同,以便共同遵守。 第一条总则 一、工程名称: 二、国家(或各部委、或省、自治区、市政府、计委、建委等有关单位)对工程的批准投资计划、工程项目表、申请建筑许可执照,计划任务书,初步设计、总概算等文件号。 三、工程编号:。

四、工程地点:。 五、工程范围:本合同全部工程建筑安装面积共计平方米(各单项工程建筑 安装面积详见工程项目一览表略)。 六、工程造价:本合同全部工程施工图预算造价为人民币元(各单项工程造 价详见工程项目一览表略)。 第二条工程期限 本合同全部工程年月日开工至年月日竣工(各单项工程、中间交工工程开、竣工日期详见工程项目一览表略)。 在组织施工过程中,如遇下列情况,得顺延工期,双方应及时进行协商,并通过书面形式确定顺延期限: 一、因天灾或人力不能抗拒的原因被迫停工者; 二、因甲方提出变更计划或变更施工图而不能继续施工者; 三、因甲方不能按期供图、供料、供设备或其所供材料,以及设备不合规格要求,被迫停工或不能顺利施工者。 第三条施工准备 一、甲方在开工前应办妥征地拆迁:申请领取建筑执照;清除施工场地范围内影响施工的原有管线、绿化等障碍物;解决施工用地(包括材料、构件的堆放和中转场地,搭建大型临时设施用地);解决施工用水源、电源和运输道路的接通;应于年月日向乙方提供所有工程设计图纸份;组织设计、施工单位进行工程设计交底。 二、乙方在开工前应组织有关人员学习和熟悉图纸,参与设计交底;编好施工图预算;负责编制施工组织设计或施工方案;进行施工场地的平整,施工地区内的用水、用电、道路以及搭建施工临时设施,安排施工总进度计划,储备材料,

最高法工程合同司法解释(二) -10 第十一条:多份合同均无效时建设工程价款结算依据-1

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工程合同_9

工程合同 工程合同篇1发包方(以下简称甲方):工程地址:住所地址:联系电话:手机号码: 承包方(以下简称乙方):单位地址:乙方代表人:工程监理及电话:本工程设计师:联系电话:本工程项目经理:联系电话: 依照《中华人民共和国合同法》及其他有关法律、法规的规定,结合家庭装饰装修的特点,甲、乙双方在平等、自愿、协商一致的基础上,就乙方承包甲方的家庭装饰装修工程(以下均以简称工程)的有关事宜,达成以下协议条款:第一条工程概况 1.1工程地点及面积;; 1.2工程造价:¥元,大写(人民币):;(若要变更施工内容、变更材料,工程造价按实结算) 1.3工程承包方方式,双方商定采取下列第种承包方式: (1)乙方包工、包全部材料; (2)乙方包工、部分包材料,甲方提供部分材料; (3)乙方包工、甲方包全部材料。 1.4工程有效期限天; 预计开工日期年月日 预计竣工日期年月日 第二条工程监理 若本工程实行工程第三方监理,甲方(或乙方)与监理公

司另行签订《工程监理合同》,并将监理工程师的姓名、单位、联系方式及监理工程师的职责等通知乙方(或甲方)。 第三条施工图纸 双方商定施工图纸采取下列第种方式提供: 3.1如甲方自行设计并提供施工图纸,交图时间为年月日,图纸为一式三份,甲方、乙方、施工队各执一份(见附件六:家庭装饰装修工程设计施工图纸); 3.2甲方委托乙方设计施工图纸,图纸一式三份,甲方、乙方、施工队各执一份(见附件六:家庭装饰装修工程设计图纸),设计费由甲方支付(此费用不在工程价款内); 3.3施工图纸双方签字后立即生效。 第四条甲方义务 4.1开工前天内,甲方应为乙方入场施工提供施工条件,全部或部分腾空房屋,清除影响施工的障碍物:对只能部分腾空的房屋中所滞留的家具、陈设应采取自行保护措施,以不影响施工为原则。乙方应对甲方施工现场的物品应认真清点及保护; 4.2开工前应提供施工期间的水源、电源,并说明使用注意事项; 4.3负责协调施工队与物业、邻里等之间的关系; 4.4禁止拆动室内承重结构,如确需拆该建筑的承重、非承重结构或迁改厨、卫位置及设备管道管线,应负责到有关部门办理相关的审批手续;根据有关部门规定执行,甲方无权要求乙方移动或改造暖气、燃气管、户外主供水供电设

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