A Holistic Survey of Wireless Multipath Video Streaming <s> 1) UDP: <s> This paper proposes a mechanism for equation-based congestion control for unicast traffic. Most best-effort traffic in the current Internet is well-served by the dominant transport protocol, TCP. However, traffic such as best-effort unicast streaming multimedia could find use for a TCP-friendly congestion control mechanism that refrains from reducing the sending rate in half in response to a single packet drop. With our mechanism, the sender explicitly adjusts its sending rate as a function of the measured rate of loss events, where a loss event consists of one or more packets dropped within a single round-trip time. We use both simulations and experiments over the Internet to explore performance. <s> BIB001 </s> A Holistic Survey of Wireless Multipath Video Streaming <s> 1) UDP: <s> In this paper, we explore end-to-end loss differentiation algorithms (LDAs) for use with congestion-sensitive video transport protocols for networks with either backbone or last-hop wireless links. As our basic video transport protocol, we use UDP in conjunction with a congestion control mechanism extended with an LDA. For congestion control, we use the TCP-Friendly Rate Control (TFRC) algorithm. We extend TFRC to use an LDA when a connection uses at least one wireless link in the path between the sender and receiver. We then evaluate various LDAs under different wireless network topologies, competing traffic, and fairness scenarios to determine their effectiveness. In addition to evaluating LDAs derived from previous work, we also propose and evaluate a new LDA, ZigZag, and a hybrid LDA, ZBS, that selects among base LDAs depending upon observed network conditions. We evaluate these LDAs via simulation, and find that no single base algorithm performs well across all topologies and competition. However, the hybrid algorithm performs well across topologies and competition, and in some cases exceeds the performance of the best base LDA for a given scenario. All of the LDAs are reasonably fair when competing with TCP, and their fairness among flows using the same LDA depends on the network topology. In general, ZigZag and the hybrid algorithm are the fairest among all LDAs. <s> BIB002 </s> A Holistic Survey of Wireless Multipath Video Streaming <s> 1) UDP: <s> The Datagram Congestion Control Protocol (DCCP) is a transport ::: protocol that provides bidirectional unicast connections of ::: congestion-controlled unreliable datagrams. DCCP is suitable for ::: applications that transfer fairly large amounts of data and that can ::: benefit from control over the tradeoff between timeliness and ::: reliability. [STANDARDS-TRACK] <s> BIB003 </s> A Holistic Survey of Wireless Multipath Video Streaming <s> 1) UDP: <s> This document contains the profile for Congestion Control Identifier 2 ::: (CCID 2), TCP-like Congestion Control, in the Datagram Congestion ::: Control Protocol (DCCP). CCID 2 should be used by senders who would ::: like to take advantage of the available bandwidth in an environment ::: with rapidly changing conditions, and who are able to adapt to the ::: abrupt changes in the congestion window typical of TCP's Additive ::: Increase Multiplicative Decrease (AIMD) congestion control. ::: [STANDARDS-TRACK] <s> BIB004 </s> A Holistic Survey of Wireless Multipath Video Streaming <s> 1) UDP: <s> This document contains the profile for Congestion Control Identifier ::: 3, TCP-Friendly Rate Control (TFRC), in the Datagram Congestion ::: Control Protocol (DCCP). CCID 3 should be used by senders that want a ::: TCP-friendly sending rate, possibly with Explicit Congestion ::: Notification (ECN), while minimizing abrupt rate changes. [STANDARDS- ::: TRACK] <s> BIB005 </s> A Holistic Survey of Wireless Multipath Video Streaming <s> 1) UDP: <s> In this paper, we propose a novel transport protocol that effectively utilizes available bandwidth and diversity gains provided by heterogeneous, highly lossy paths. Our Multi-Path LOss-Tolerant (MPLOT) protocol can be used to provide significant gains in the goodput of wireless mesh networks, subject to bursty, correlated losses with average loss-rates as high as 50%, and random outage events. MPLOT makes intelligent use of erasure codes to guard against packets losses, and a Hybrid-ARQ/FEC scheme to reduce packet recovery latency, where the redundancy is adaptively provisioned into both proactive and reactive FECs. MPLOT uses dynamic packet mapping based on current path characteristics, and does not require packets to be delivered in sequence to ensure reliability. We present a theoretical analysis of the different design choices of MPLOT and show that MPLOT makes an optimal trade-off between goodput and delay constraints. We test MPLOT, through simulations, under a variety of test scenarios and show that it effectively exploits path diversity in addition to aggregating path bandwidths. We also show that MPLOT is fair to single-path protocols like TCP-SACK. <s> BIB006 </s> A Holistic Survey of Wireless Multipath Video Streaming <s> 1) UDP: <s> Multipath streaming protocols have recently attracted much attention because they provide an effective means to provide high-quality streaming over the Internet. However, many existing schemes require a long start-up delay and thus are not suitable for interactive applications such as video conferencing and tele-presence. In this paper, we focus on real-time live streaming applications with stringent end-to-end latency requirement, say several hundreds of milliseconds. To address these challenges, we take a joint multipath and FEC approach that intelligently splits the FEC-encoded stream among multiple available paths. We develop an analytical model and use asymptotic analysis to derive closed-form, optimal load splitting solutions, which are surprisingly simple yet insightful. To our best knowledge, this is the first work that provides such closed-form optimal solutions. Based on the analytical insights, we have designed and implemented a novel Encoded Multipath Streaming (EMS) scheme for real-time live streaming. EMS strives to continuously satisfy the application's QoS requirements by dynamically adjusting the load splitting decisions and the FEC settings. Our simulation results have shown that EMS can not only outperform the existing multipath streaming schemes, but also adapt to the dynamic loss and delay characteristics of the network with minimal overhead. <s> BIB007 </s> A Holistic Survey of Wireless Multipath Video Streaming <s> 1) UDP: <s> New emerging technologies like Multimedia applications require timely delivery of information as compared to reliability. The most widely used protocols over transport layer are Transmission Control Protocol (TCP) and User Datagram Protocol (UDP). Both of these have some drawbacks while using in real time applications. TCP increases delay in transfer of data but achieving reliable transfer of data while UDP does not provide any acknowledgment, hence it overcomes the problem of delay but lacking congestion control mechanism. To provide congestion control over transport layer using UDP, Datagram Congestion Control Protocol (DCCP) has been designed to provide timely delivery of data and congestion control mechanism as well. In this research text, we have worked out the behavior of video applications over DCCP environment. We have analyzed the performance of MPEG4 video over transport layer using DCCP variants, UDP and TCP under competing and non-competing topology. Delay, throughput, jitter and packet loss are used to assess the performance of these protocols. Our results show that DCCP can be used as a transport layer protocol for video applications, which may result in improved throughput, reduced packet loss as compared to UDP but the delay and jitter of UDP is much lesser than DCCP variants. According to the results DCCP assures better QoS than other transport protocols for video transmission under congestion. <s> BIB008 </s> A Holistic Survey of Wireless Multipath Video Streaming <s> 1) UDP: <s> Datagram Congestion Control Protocol (DCCP) is an unreliable transport layer protocol with a congestion control mechanism used for multimedia streaming. In order to use multiple network interfaces to transmit streaming data smoothly, a Multi-Path Datagram Congestion Control Protocol (MP-DCCP) is proposed and presented in this paper. Video streaming transmission through multi-path faces three problems: (1) out-of-ordering packets at the receiver side, (2) conditions of paths are changed anytime, and (3) the importances of frames/packets are different. The first problem may let video streaming data be delivered to the application layer too late, especially for the live streaming data. The second and third problems may let a packet be scheduled to an unsuitable path for transmission. Since importances of video streaming frames/packets are different, it should consider how to schedule the transmission of frames/packets properly. For example, an important frame/packet should be transmitted in the more reliable path. In order to resolve these three problems, a QoS-aware Order Prediction Scheduling (QOPS) scheme for MP-DCCP is proposed in this paper. QOPS estimates packets' arrival orders at the receive side through these multiple paths before packets are scheduled into the transmission at the sender side. From simulation results, we proved that the packets' out-of-ordering problem in MPDCCP can be improved using our proposed QOPS scheme. <s> BIB009 </s> A Holistic Survey of Wireless Multipath Video Streaming <s> 1) UDP: <s> This paper proposes a robust video streaming scheme for priority networks with path diversity, based on a combined approach of multiple description coding (MDC) with optimal picture classification into two priorities. A binary classification algorithm is proposed to define high (HP) and low (LP) priority network abstraction layer units (NALU), which in turn define the packet priorities. An optimisation algorithm is used to find HP pictures, based on dynamic programming and relying on minimisation of the packet loss concealment distortion. The paper shows that the proposed algorithm is able to effectively improve the decoded video without increasing the MDC stream redundancy. The overall performance evaluation, carried out by simulating MDC video streaming over lossy networks with path diversity, demonstrates that the proposed algorithm yields higher video quality for a wide range of packet loss rates (PLR). Comparing with no-priority MDC video streaming schemes, the simulation results show that the proposed algorithm can improve the average PSNR results in 0.7–3.2dB for packet loss rates between 3% and 15%. <s> BIB010 </s> A Holistic Survey of Wireless Multipath Video Streaming <s> 1) UDP: <s> We consider the problem of scalable video streaming from a server to multinetwork clients over heterogeneous access networks, with the goal of minimizing the distortion of the received videos. This problem has numerous applications including: 1) mobile devices connecting to multiple licensed and ISM bands, and 2) cognitive multiradio devices employing spectrum bonding. In this paper, we ascertain how to optimally determine which video packets to transmit over each access network. We present models to capture the network conditions and video characteristics and develop an integer program for deterministic packet scheduling. Solving the integer program exactly is typically not computationally tractable, so we develop heuristic algorithms for deterministic packet scheduling, as well as convex optimization problems for randomized packet scheduling. We carry out a thorough study of the tradeoff between performance and computational complexity and propose a convex programming-based algorithm that yields good performance while being suitable for real-time applications. We conduct extensive trace-driven simulations to evaluate the proposed algorithms using real network conditions and scalable video streams. The simulation results show that the proposed convex programming-based algorithm: 1) outperforms the rate control algorithms defined in the Datagram Congestion Control Protocol (DCCP) by about 10-15 dB higher video quality; 2) reduces average delivery delay by over 90% compared to DCCP; 3) results in higher average video quality of 4.47 and 1.92 dB than the two developed heuristics; 4) runs efficiently, up to six times faster than the best-performing heuristic; and 5) does indeed provide service differentiation among users. <s> BIB011 </s> A Holistic Survey of Wireless Multipath Video Streaming <s> 1) UDP: <s> With bounded receive buffers, the aggregate bandwidth of multipath transmission degrades significantly in the presence of path heterogeneity. The performance could even be worse than that of single-path TCP, undermining the advantage gained by using multipath transmit. Furthermore, multipath transmission also suffers from delay and jitter even with large receive buffers. In order to tolerate the path heterogeneity when the receive buffer is bounded, we propose a new multipath TCP protocol, namely SC-MPTCP, by integrating linear systematic coding into MPTCP. In SC-MPTCP, we make use of coded packets as redundancy to counter against expensive retransmissions. The redundancy is provisioned into both proactive and reactive data. Specifically, to send a generation of packets, SC-MPTCP transmits proactive redundancy first and then delivers the original packets, instead of encoding all sent-out packets as all the existing coding solutions have done. The proactive redundancy is continuously updated according to the estimated aggregate retransmission ratio. In order to avoid the proactive redundancy being underestimated, the pre-blocking warning mechanism is utilized to retrieve the reactive redundancy from the sender. We use an NS-3 network simulator to evaluate the performance of SC-MPTCP with and without the coupled congestion control option. The results show that with bounded receive buffers, MPTCP achieves less than 20% of the optimal goodput with diverse packet losses, whereas SC-MPTCP approaches the optimal performance with significantly smaller receive buffers. With the help of systematic coding, SC-MPTCP reduces the average buffer delay of MPTCP by at least 80% in different test scenarios. We also demonstrate that the use of systematic coding could significantly reduce the arithmetic complexity compared with the use of non-systematic coding. <s> BIB012 </s> A Holistic Survey of Wireless Multipath Video Streaming <s> 1) UDP: <s> Video streaming currently dominates global Internet traffic and will be of even increasing importance in the future. In this paper we assess the impact of the underlying transport protocol on the user perceived quality for video streaming using YouTube as example. In particular, we investigate whether UDP or TCP fits better for Video-on-Demand delivery from the end user's perspective, when the video is transmitted over a bottleneck link. For UDP based streaming, the bottleneck link results in spatial and temporal video artifacts, decreasing the video quality. In contrast, in the case of TCP based streaming, the displayed content itself is not disturbed but playback suffers from stalling due to rebuffering. The results of subjective user studies for both scenarios are analyzed in order to assess the transport protocol influences on Quality of Experience of YouTube. To this end, application-level measurements are conducted for YouTube streaming over a network bottleneck in order to develop models for realistic stalling patterns. Furthermore, mapping functions are derived that accurately describe the relationship between network-level impairments and QoE for both protocols. <s> BIB013 </s> A Holistic Survey of Wireless Multipath Video Streaming <s> 1) UDP: <s> Multipathing in communication networks is gaining momentum due to its attractive features of increased reliability, throughput, fault tolerance, and load balancing capabilities. In particular, wireless environments and datacenters are envisioned to become largely dependent on the power of multipathing for seamless handovers, virtual machine (VM) migration and in general, pooling less proficient resources together for achieving overall high proficiency. The transport layer, with its knowledge about end-to-end path characteristics, is well placed to enhance performance through better utilization of multiple paths. Realizing the importance of transport-layer multipath, this paper investigates the modernization of traditional connection establishment, flow control, sequence number splitting, acknowledgement, and flow scheduling mechanisms for use with multiple paths.Multipath rate control defines the fundamental feature of the transport layer. Thus, we study the restructuring of classical utility maximization framework and analyze its stability and convergence. We also examine the topic of multipath congestion control in the light of TCP fairness. To the best of our knowledge, this is the first in-depth survey paper that has chronicled the evolution of the transport layer of the Internet from the traditional single-path TCP to the recent development of the modern multipath TCP (MPTCP) protocol. Along with describing the history of this evolution, we also highlight in this paper the remaining challenges and research issues. Graphical abstractDisplay Omitted <s> BIB014 </s> A Holistic Survey of Wireless Multipath Video Streaming <s> 1) UDP: <s> Recent technological advancements in wireless infrastructures and handheld devices enable mobile users to concurrently receive multimedia contents with different radio interfaces (e.g., cellular and Wi-Fi). However, multipath video transport over the resource-limited and error-prone wireless networks is challenged with key technical issues: 1) conventional multipath protocols are throughput-oriented, and video data are scheduled in a content-agnostic fashion and 2) high-quality real-time video is bandwidth-intensive and delay-sensitive. To address these critical problems, this paper proposes a bandwidth-efficient multipath streaming (BEMA) protocol featured by the priority-aware data scheduling and forward error correction-based reliable transmission. First, we present a mathematical framework to formulate the delay-constrained distortion minimization problem for concurrent video transmission over multiple wireless access networks. Second, we develop a joint Raptor coding and data distribution framework to achieve target video quality with minimum bandwidth consumption. The proposed BEMA is able to effectively mitigate packet reordering and path asymmetry to improve network utilization. We conduct performance evaluation through extensive emulations in Exata involving real-time H.264 video streaming. Compared with the existing multipath protocols, BEMA achieves appreciable improvements in terms of video peak signal-to-noise ratio, end-to-end delay, bandwidth utilization, and goodput. Therefore, BEMA is recommended for streaming high-quality real-time video to multihomed terminals in heterogeneous wireless networks. <s> BIB015 </s> A Holistic Survey of Wireless Multipath Video Streaming <s> 1) UDP: <s> This document describes, surveys, and classifies the protocol ::: mechanisms provided by existing IETF protocols, as background for ::: determining a common set of transport services. It examines the ::: Transmission Control Protocol (TCP), Multipath TCP, the Stream Control ::: Transmission Protocol (SCTP), the User Datagram Protocol (UDP), UDP- ::: Lite, the Datagram Congestion Control Protocol (DCCP), the Internet ::: Control Message Protocol (ICMP), the Real-Time Transport Protocol ::: (RTP), File Delivery over Unidirectional Transport / Asynchronous ::: Layered Coding (FLUTE/ALC) for Reliable Multicast, NACK- Oriented ::: Reliable Multicast (NORM), Transport Layer Security (TLS), Datagram ::: TLS (DTLS), and the Hypertext Transport Protocol (HTTP), when HTTP is ::: used as a pseudotransport. This survey provides background for the ::: definition of transport services within the TAPS working group. <s> BIB016 </s> A Holistic Survey of Wireless Multipath Video Streaming <s> 1) UDP: <s> HTTP adaptive streaming (HAS) is the most recent attempt regarding video quality adaptation. It enables cheap and easy to implement streaming technology without the need for a dedicated infrastructure. By using a combination of TCP and HTTP it has the advantage of reusing all the existing technologies designed for ordinary web. Equally important is that HAS traffic passes through firewalls and works well when NAT is deployed. The rate adaptation controller of HAS, commonly called adaptive bitrate selection (ABR), is currently receiving a lot of attention from both industry and academia. However, most of the research efforts concentrate on a specific aspect or a particular methodology without considering the overall context. This paper presents a comprehensive survey of the most significant research activities in the area of client-side HTTP-based adaptive video streaming. It starts by decomposing the ABR module into three subcomponents, namely: resource estimation function, chunk request scheduling, and adaptation module. Each subcomponent encapsulates a particular function that is vital to the operation of an ABR scheme. A review of each of the subcomponents and how they interact with each other is presented. Furthermore, those external factors that are known to have a direct impact on the performance of an ABR module, such as content nature, CDN, and context, are discussed. In conclusion, this paper provides an extensive reference for further research in the field. <s> BIB017 </s> A Holistic Survey of Wireless Multipath Video Streaming <s> 1) UDP: <s> This memo updates RFC 3168, which specifies Explicit Congestion ::: Notification (ECN) as an alternative to packet drops for indicating ::: network congestion to endpoints. It relaxes restrictions in RFC 3168 ::: that hinder experimentation towards benefits beyond just removal of ::: loss. This memo summarizes the anticipated areas of experimentation ::: and updates RFC 3168 to enable experimentation in these areas. An ::: Experimental RFC in the IETF document stream is required to take ::: advantage of any of these enabling updates. In addition, this memo ::: makes related updates to the ECN specifications for RTP in RFC 6679 ::: and for the Datagram Congestion Control Protocol (DCCP) in RFCs 4341, ::: 4342, and 5622. This memo also records the conclusion of the ECN nonce ::: experiment in RFC 3540 and provides the rationale for reclassification ::: of RFC 3540 from Experimental to Historic; this reclassification ::: enables new experimental use of the ECT(1) codepoint. <s> BIB018
The User Datagram Protocol (UDP)  , standardized by IETF in 1980, is widely used for unidirectional, broadcast, unicast, multicast, and anycast communications. Next, we provide a brief recap of UDP basics and discuss relevant multipath efforts. UDP overview. UDP was designed to use a single path for data transmission. It is a connectionless protocol, it does not use sequence numbers for data transmission BIB014 , and there is no guarantee for in-ordered and reliable delivery. UDP also has no congestion control for bandwidth adaptation. These properties make UDP a fast transmission protocol BIB016 upon which video streaming solutions can be easily implemented. However, the lake of bandwidth adaptation causes UDP to transmit the data with the same bit rate as sent by the application. Therefore, when the network is congested, unless the application holds back, packets get discarded leading to video distortion and reduced QoE BIB013 . Moreover, without congestion control, UDP may occupy a high fraction of the available bandwidth, and consequently, acting unfair to other congestion-avoiding network flows BIB009 . Multipath support. There are several efforts to add multipath transmission and bandwidth aggregation to UDP for video streaming BIB015 , BIB011 , BIB010 . Note that the approaches proposed in BEMA BIB015 and BIB011 introduced rate balancing methods to avoid network congestion. Wu et al. BIB015 designed a Bandwidth-Efficient Multipath streAming (BEMA) protocol and claimed that it was the first work that employed Raptor coding and priority-aware scheduling to stream HD real-time video over heterogeneous wireless networks. This content-aware model sends packets with higher priority on the better-qualified paths and I-frame packets through all available paths. The approach also utilizes Forward Error Correction (FEC) to protect transmission data. BEMA also provides a TCP-Friendly Rate Control (TFRC) in order to guarantee fairness toward TCP flows. TFRC BIB001 is an equation-based congestion control algorithm, which is designed for unicast multimedia traffic. TFRC estimates the loss event rate at receiver and informs it to the sender, which adapts its transmission rate based on the congestion estimation and on the equation that models TCP congestion control behavior. TFRC responds to the congestion with less fluctuation than standard TCP congestion control and over longer periods of time BIB002 . However, TFRC may cause unnecessary reduction of transmission rate during wireless losses. BEMA then adds a ZigZag scheme BIB002 in order to distinguish congestion losses from wireless losses. Only if ZigZag classifies a packet loss as a congestion loss, TFRC will consider it as a lost packet BIB002 . Considering the relevance of the feedback information for the proper scheduling process and its high effect on the performance, it is sent periodically from the client to the server over a reliable TCP connection. Freris et al. BIB011 proposed a distortion-aware scalable video streaming to multiple multihomed clients. The authors claimed that their work is the first that simultaneously considered End-to-End rate control and scalable stream adaptation for multipath over heterogeneous access networks. In this approach, the requested video stream is divided into substreams on the server side. The authors developed an algorithm to determine the rate of each substream and the packets to be included in each substream considering network information (e.g., available bandwidth and RTT) and video content features in order to minimize video distortion. Besides that, different cost functions are proposed to provide service differentiation and fairness among users. The authors also developed heuristic algorithms for deterministic packet scheduling. Once it is a scalable streaming approach, each packet is transmitted only if all other related packets in lower layers have been sent before. Substreams integrate into a single scalable video stream at the client. The authors also studied the trade-off between performance and computational complexity and concluded that it works better for a small number of clients because of overhead. Correia et al. BIB010 proposed a video streaming approach for networks with path diversity using MDC as an error resilience technique. The authors proposed a priority classification. A limited number of packets were classified as high priority because they minimize the distortion of the decoded video affected by packet loss. These packets are delivered without losses. Remaining low priority packets are prone to transmission losses. 2) TCP: Transmission Control Protocol (TCP)  is a transport protocol standardized by IETF in 1981. This protocol has been widely adopted for video streaming in Real-Time Communications (RTC)  and in HTTP-based applications. We previously discussed TCP lack of throughput stability BIB015 with its negative effect on adaptive bit streaming in Section IV-A2. Here, we provide more details about TCP and discuss one surveyed work that is based on this protocol. TCP overview. TCP is designed to use a single path for data transmission. Regarding data transmission process, TCP uses sequence numbers to detect losses, guarantee in-order packet delivery, and reconstruct the received data BIB014 . The receiver sends ACKs for the correctly received packets. These ACKs are used to provide reliable communication. Retransmission occurs in two cases. First, when there is no ACK from the receiver, which is detected by using a retransmission timer referred to as Retransmission Time-Out (RTO). Second, when the sender receives three duplicate ACKs, which means loss occurred. As previously also discussed in Section I, retransmission wastes bandwidth and adds significant delays. Several protocol improvements have been proposed. For example, Selective Acknowledgements (SACK)  , where the receiver informs the sender all successfully arrived packets, so the sender retransmits only the segments that have actually been lost, and Cumulative ACK, which acknowledge the last successfully received packet to the sender. In addition, Explicit Congestion Notification (ECN) BIB018 has been proposed as an optional capability to collect congestion information hop by hop and inform the sender about the congestion levels. Using congestion control by monitoring packet losses and/or delay variations BIB014 , TCP enables to adapt the data rate to network congestion and leads to minimize packet loss BIB013 . In case of not enough network bandwidth available, TCP sends video data with a lower bit rate than the required video bit rate. Thus, video transmission takes longer than the video playback, and consequently may cause the playback to stall. While stall has a severe effect on the perceived video quality, in case of VoD delivery, typically, stall is preferred over video distortions BIB013 . Previously in Section III, we explained about HOL issues and liveness strategies used in TCP-based applications for live or interactive video streaming to cope with stall and delay constraints requirements. Besides all the explained properties, TCP has also the advantage of traversing through firewalls and NATs, a common issue in UDP, altogether turning TCP into a dominant transport protocol for video services BIB017 . Multipath support. Sharma et al. BIB006 proposed MultiPath LOss-Tolerant (MPLOT) protocol based on SACK-based TCP and cumulative ACK. A framework, named Hybrid-ARQ (HARQ)/FEC, is defined for MPLOT. Based on HARQ/FEC, MPLOT is using adaptive FEC proactively and reactively instead of high retransmissions to recover losses. Proactive FEC (PFEC) packets are used to recover losses and when PFEC packets in a block are not enough to recover lost data, then Reactive FEC (RFEC) packets need to transmit. This method leads to goodput improvement and decreased recovery latency in high lossy channels BIB007 . Regarding packet scheduling, paths in MPLOT are categorized into good and bad paths. The channels with ranks higher than a threshold (median rank) are categorized as good paths. Ranks are calculated based on network parameters, such as congestion window, PLR and RTT. MPLOT provides an uncoupled congestion control which means each path has its own congestion control. ECN is used to find congestion losses (from faulty/lost channels) and to change the congestion window size. However, MPLOT is deployed for wireless mesh networks and it is not easily expendable on the Internet due to scalability and compatibility issues. The authors assume that a buffer is enough to compensate out of order delivered packets, which are important in video quality BIB007 , BIB012 . Moreover, the approach is using a CBR coding scheme, which decreases the performance when the path quality decreases sharply BIB007 . 3) DCCP: Datagram Congestion Control Protocol (DCCP) BIB003 is a transport protocol standardized by IETF in 2006. Here, firstly, we provide an overview of DCCP, such as data transmission process, and its properties. Then, we discuss one surveyed work that is based on this protocol. DCCP overview. DCCP is designed to use a single path for data transmission providing bidirectional and unicast data delivery. Regarding data transmission process, DCCP uses sequence numbers. Therefore, the client can detect losses and inform them to the sender by ACKs. There is no retransmission method and in-order data delivery. In addition, there is an ability for feature negotiation before or during transmission, such as ECN capability, ACK ratio, and congestion control mechanism. DCCP has different congestion control mechanisms that are represented by Congestion Control IDentifier (CCID), for example, CCID2 and CCID3. CCID2 has a TCP-like Congestion Control. Thus, the sender has a congestion window and sends data until making the window full. Both dropped packets and ECN trigger the congestion algorithm and halve the congestion window. Acknowledgments contain a list of received packets within some window, like Selective Acknowledgements (SACK)-based TCP. Therefore, CCID2 BIB004 provides quick access to available bandwidth and deals with quick bit rate changing BIB003 , BIB009 . CCID3 BIB005 provides TFRC. CCID3 responds to congestion smoothly and maintains steady bit rate BIB003 , BIB009 . A comparison among UDP, TCP and DCCP variants (CCID2 and CCID3) for transferring MPEG4 video, shows that DCCP provides higher throughput and less packet loss compared to UDP while UDP supplies much less delay and jitter. Finally, DCCP comes up with the best QoS compared with TCP and UDP transport protocols over congested network BIB008 . However, since subjective results in the work BIB013 shows stalling caused by TCP is preferred over distortion caused by UDP for VoD streaming, DCCP without retransmission may also suffer from video distortion and may not outperform TCP and UDP for VoD in terms of QoE.