Abstract:
Systems and methods for communication of scaleable-coded audiovisual signals over multiple TCP/IP connections are provided. The sender schedules and prioritizes transmission of individual scalable-coded data packets over the plurality of TCP connections according to their relative importance in the scalable coding structure for signal reconstruction quality and according to receiver feedback. Low-latency packet delivery over the multiple TCP/IP connections is maintained by avoiding transmission or retransmission of packets that are less important for reconstructed media quality.

Description:
CROSS-REFERENCE TO RELATED APPLICATIONS 
       [0001]    This application is a continuation in part of International Application Serial No. PCT/US06/028365, filed Jul. 20, 2006, which claims priority from U.S. Provisional Patent Application No. 60/701,108 filed Jul. 20, 2005; a continuation in part of International patent application No. PCT/US06/028366 filed Jul. 20, 2006 which claims priority from U.S. Provisional Patent Application No. 60/701,109 filed Jul. 20, 2005; a continuation in part of International patent application No. PCT/US06/061815 filed Dec. 8, 2006, which claims priority from U.S. Provisional Patent Application No. 60/748,437 filed Dec. 8, 2005; a continuation in part of International patent application No. PCT/US06/062569 filed Dec. 22, 2006, which claims priority from United States Provisional Patent Application No. 60/753,343 filed Dec. 22, 2005; and a continuation in part of International patent application No. PCT/US07/63335 filed Mar. 5, 2007 which claims priority from U.S. Provisional Patent Application No. 60/778, 760 filed Mar. 3, 2007; and a continuation in part of International patent application No. PCT/US07/083351 filed Nov. 1, 2007. All of the aforementioned applications, which are commonly assigned, are hereby incorporated by reference herein in their entireties. 
     
    
     FIELD OF THE INVENTION 
       [0002]    The present invention relates to low-delay, interactive communication systems. In particular, the invention relates to achieving low latency in packet-based communication systems in which multiple Transmission Control Protocol (TCP) connections are used for transmitting scalable coded data. 
       BACKGROUND OF THE INVENTION 
       [0003]    The Transmission Control Protocol (TCP) is a transport layer protocol for reliable delivery of Internet (IP) packets (datagrams). TCP uses an Additive Increase Multiplicative Decrease (AIMD) rate control mechanism to ensure fair use of shared network resources (e.g., the available bit rate). With TCP/AIMD operation, whenever all outstanding packets sent within the last round-trip time (RTT) cycle are acknowledged by the receiver, TCP increases the transmission rate of the sender by a constant amount additively. On the other hand, when TCP detects congestion (or packet loss) by not having all outstanding packets acknowledged by the onset of the next RTT period, it halves the transmission rate of the sender, i.e., it multiplicatively reduces the rate by a factor of ½. Such TCP/AIMD rate control operation can create significant variations in the transmission bit rates, leading to exceedingly high latencies in packet delivery. This drawback makes TCP unsuitable for transport of interactive media packets, which are typically characterized by stringent delivery deadlines. 
         [0004]    In some situations involving interactive multimedia communications, however, it is necessary to employ TCP transport in spite of its drawbacks. For example, corporate firewalls are sometimes set to block all traffic to, and from, the corporate Local Area Network (LAN) except over TCP connections. Therefore, media packets from the outside world destined for a receiver on the corporate LAN must be delivered via TCP, or otherwise face the prospect of being blocked by the firewall prior to entering the LAN. 
         [0005]    Several studies or investigations on the use of TCP for interactive media transmission have been reported. See, e.g., Sally Floyd, Mark Handley, Jitendra Padhye, and Joerg Widmer, “Equation-Based Congestion Control for Unicast Applications,” August 2000, SIGCOMM 2000; Bing Wang, Wei Wei, Zheng Guo, and Don Towsley, “Multipath Live Streaming via TCP: Performance and Benefits,” UConn CSE Technical Report: BECAT/CSE-TR-06-7; S. Sakazawa, Y. Takishima, Y. Nakajima, M. Wada, and K. Hashimoto, “Multimedia contents management and transmission system ‘VAST-web’ and its effective transport protocol ‘SVFTP’”, ICME 2004; and T. Nguyen and S.-C. Cheung, “Multimedia Streaming Using Multiple TCP Connections,” IPCCC 2005. 
         [0006]    The first of these studies (i.e., Equation-Based Congestion Control for Unicast Applications) describes a TCP-friendly scheme, which provides an equation-based rate control technique as an alternative to the TCP/AIMD rate control mechanism while preserving the feature of sharing in a fair manner the available network bit rate with existing TCP flows. The equation-based rate control technique yields smoother send rate fluctuations (than TCP/AIMD) in response to network congestion, and therefore makes it more suitable for streaming applications. The second of the cited studies (i.e., Multipath Live Streaming via TCP: Performance and Benefits) considers employing TCP transport over multiple network paths in order to improve TCP performance for streaming applications. Similarly, the third and fourth of the cited studies (i.e., Multimedia contents management and transmission system ‘VAST-web’ and its effective transport protocol ‘SVFTP’, and Multimedia Streaming Using Multiple TCP Connections, respectively) explore transmission over multiple TCP connections on the same network path as a way to increase TCP throughput in media streaming. These two studies, however, deal only with stored (pre-encoded) media content in the context of multimedia content management systems and streaming applications, respectively; furthermore, they treat the individual media packets uniformly, and do not take advantage of a possible scalable structure in the transmitted media. When scalable coding is used in the transmitted media, different packets have different importance in terms of how they affect the reconstruction quality of the media in the receiver. 
         [0007]    Scalable coding is a well-known technique in multimedia data encoding, in which the encoder generates two or more “scaled” bitstreams that collectively represent a given medium in a bandwidth-efficient manner. Scalability can be provided in a number of different dimensions, namely temporal, spatial, and quality (also referred to as SNR (Signal-to-Noise Ratio) scalability) dimensions. For example, a video signal may be scalable-coded in different layers at CIF and QCIF resolutions, and at frame rates of 7.5, 15, and 30 frames per second (fps). Depending on the codec&#39;s structure, any combination of spatial resolutions and frame rates may be obtainable from the codec bitstream. The bits corresponding to the different layers can be transmitted as separate bitstreams (i.e., one stream per layer), or they can be multiplexed together in one or more bitstreams. For convenience in description herein, the coded bits corresponding to a given layer may be referred to as that layer&#39;s bitstream, even if the various layers are multiplexed and transmitted in a single bitstream. Codecs specifically designed to offer scalability features include, for example, MPEG-2 (ISO/IEC 13818-2, also known as ITU-T H.262) and the currently developed SVC (known as ITU-T H.264 Annex G or MPEG-4 Part 10 SVC). Scalable coding techniques specifically designed for video communication are described, for example, in commonly assigned International Patent Application No. PCT/US06/028365 “SYSTEM AND METHOD FOR SCALABLE AND LOW-DELAY VIDEOCONFERENCING USING SCALABLE VIDEO CODING.” 
         [0008]    It is noted that even codecs that are not specifically designed to offer scalability features can exhibit scalability characteristics in the temporal dimension. For example, consider an MPEG-2 Main Profile codec, a non-scalable codec, which is used in DVDs and digital TV environments. Further, assume that the codec is operated at 30 fps and that a group of pictures (GOP) structure of IBBPBBPBBPBBPBB (period N=15 frames) is used. By sequential elimination of the B pictures, followed by elimination of the P pictures, it is possible to derive a total of three temporal resolutions: 30 fps (all picture types included), 10 fps (I and P only), and 2 fps (I only). The sequential elimination process results in a decodable bitstream because the MPEG-2 Main Profile codec is so designed that coding of the P pictures does not rely on the B pictures, and, similarly, coding of the I pictures does not rely on other P or B pictures. For convenience, in the following description, single-layer codecs with temporal scalability features are considered to be a special case of scalable video codecs, and understood to be included in the term “scalable video coding” unless explicitly indicated otherwise. 
         [0009]    Scalable codecs typically have a pyramidal bitstream structure in which one of the constituent bitstreams (called the “base layer”) is essential in recovering the original medium at some basic quality. Use of one or more of the remaining bitstream(s) (called the “enhancement layer(s)”) together with the base layer increases the quality of the recovered medium. Data losses in the enhancement layers may be tolerable, but data losses in the base layer can cause significant distortions or complete loss of the recovered medium. 
         [0010]    Simulcasting is a coding solution that is less complex than scalable coding but has some of the advantages of the latter. In simulcasting, two different versions of the source are encoded (e.g., at two different spatial resolutions) and transmitted. Each version is independent, in that its decoding does not depend on reception of the other version. In the following description, simulcasting is considered to be a special case of scalable coding (where no inter layer prediction is performed), and referred to simply as scalable coding unless explicitly indicated otherwise. 
         [0011]    Consideration is now being given to improving packet-based communication systems in which multiple TCP connections are used for transmitting scalable coded data. In particular, attention is being directed to live audio and video communication scenarios where providing low latency packet delivery is essential. 
       SUMMARY OF THE INVENTION 
       [0012]    Systems and methods for packet-based communication of scalable coded media are provided. The systems and methods include mechanisms for TCP-based transport of media packets for low-delay, interactive communication applications such as videoconferencing. Multiple TCP connections are established between sender and receiver for communication of the media packets. The sender makes scheduling decisions based on the media packets&#39; importance in the scalable coding structure and on feedback from the receiver (e.g., on the status of individual TCP connections). 
         [0013]    The systems and methods take into account the varying importance of the scalable coded packets to the quality of the reconstructed media when making scheduling decisions. Such decisions are made to maintain low latency packet delivery and to provide an acceptable audio-visual presentation experience of the received media despite the TCP rate control mechanism. The systems and methods overcome the limitations TCP and its AIMD rate control mechanism that cause detrimental delay in interactive media applications. 
     
    
     
       BRIEF DESCRIPTION OF THE DRAWINGS 
         [0014]      FIG. 1  is a schematic diagram illustrating the architecture of a typical TCP-based communication system (prior art); 
           [0015]      FIG. 2  is a schematic diagram illustrating the architecture of an exemplary communication system having multiple TCP connections, a Scheduling Inverse Multiplexer (S-IMUX), and a Feedback Multiplexer (F-MUX), in accordance with the principles of the present invention; 
           [0016]      FIG. 3  is a schematic diagram illustrating the operation of the Scheduling Inverse Multiplexer (S-IMUX), in accordance with the principles of the present invention; 
           [0017]      FIG. 4  is a schematic diagram illustrating the operation of the Feedback Multiplexer (F-MUX), in accordance with the principles of the present invention; and 
           [0018]      FIG. 5  is a schematic diagram illustrating an exemplary scalable video picture coding structure (prior art). 
       
    
    
       [0019]    Throughout the figures the same reference numerals and characters, unless otherwise stated, are used to denote like features, elements, components or portions of the illustrated embodiments. Moreover, while the present invention will now be described in detail with reference to the figures, it is being done so in connection with the illustrative embodiments. 
       DETAILED DESCRIPTION OF THE INVENTION 
       [0020]      FIG. 1  shows the architecture of a conventional system  100  for TCP-based transmission of media data between a sender  110  and a receiver  120  over a network  130 . At sender  110 , an encoder  112  produces audio or video data that is directly provided to a TCP stack (e.g., TCP/IP stack  114 ), which then transmits packet-data through a Network Interface Controller (NIC)  116  over network  130  to receiver  120 . NIC  126  in receiver  120  receives the packet-data and provides the data via TCP/IP stack  124  to the receiver&#39;s decoder  122  for decoding and display. The TCP/IP components of sender  110  and receiver  120 , in coordination, ensure reliable delivery of the transmitted data by performing retransmission, and also apply the TCP flow control and congestion avoidance algorithms. It is noted that the connections at sender  110 , from encoder  112  to TCP/IP stack  114 , and at receiver  120 , from TCP/IP stack  124  to decoder  122 , are both unidirectional connections. The connections between the TCP/IP components and the NICs are all bi-directional, as TCP is an inherently bi-directional transport protocol (i.e., since TCP acknowledgment packets are transmitted from the Receiver to the Sender, as per standard TCP operation). 
         [0021]    Although  FIG. 1  shows the architecture of system  100  in the context of live audio or video transmission, it is to be understood that the same architecture can be used for low-delay transmission of pre-coded data. In such case, encoder  112  is replaced by a component that obtains pre-coded data from mass storage, random access memory, or another suitable digital memory device. Similarly, although  FIG. 1  shows system  100  having a one-way sender-receiver connection, it is to be understood that the system architecture shown is readily extended to two-way (interactive) communication. In such case, encoders  112  and decoders  116  are placed in symmetric positions in both the receiver and sender. 
         [0022]      FIG. 2  shows the architecture of an inventive system  200  in which sender  210  makes scheduling decisions based on the transmitted media packets&#39; importance and feedback from receiver  222 , In system  200 , sender  210  has an encoder  212 , which is a scalable encoder that is connected to a Scheduling Inverse Multiplexer (S-IMUX)  218 . S-IMUX  218  demultiplexes scalable coded?? packets received from encoder  212  over a plurality of TCP connections to TCP/IP stack  114 . For clarity,  FIG. 2  shows only two such connections in system  200  (i.e., Conn.  0  and Conn.  1 ). It will be understood, however, that any suitable number of TCP connections may be used. S-IMUX  218  is responsible for managing the transmission of the different encoded media packets over the plurality of TCP connections, taking into account the system&#39;s state as well as each packet&#39;s priority with respect to its role or hierarchical position in the scalability structure of encoder  212 . The encoded media packets are transported via TCP/IP stack  114  through NIC  112  over communication network  130  to receiver  220 . At receiver  220 , packets received through NIC  122  and TCP/IP stack  116  over the multiple TCP connections Conn.  0  and Conn.  1  are collected by a Feedback Multiplexer (F-MUX) 228. F-MUX  228  forwards the packets to decoder  222  for decoding and display or playback, and also produces the receiver feedback packets that are required in the operation of S-IMUX  218  at sender  210 . 
         [0023]    It is noted that the connections between S-DMUX  218  and the TCP/IP component  114  in sender  210 , and TCP/IP component  116  and F-MUX  228  in receiver  220  are both bi-directional. This is because application-level feedback packets are transmitted from receiver  220  to sender  210 , in addition to, and separately from, the TCP acknowledgement packets. 
         [0024]    Like system  100  shown in  FIG. 1 , system  200  is shown in  FIG. 2  in the context of live audio or video transmission. However, as in the case of system  100 , the shown system  200  architecture can be used for low-delay transmission of pre-coded data. In addition, although a one-way connection is shown, the shown system  200  architecture is readily extended to two-way (interactive) communication by duplicating the sender modules in the receiver and vice versa. 
         [0025]    The inventive system  200  differs fundamentally from conventional systems (e.g., system  100 ) in at least two ways. First, instead of establishing a single TCP connection, the inventive system transmits the media packets over multiple TCP connections ( FIGS. 2-4 ). This allows a more constant transmission rate to be maintained than is possible over a single TCP connection, and provides greater flexibility in responding to network-induced effects such as packet loss and congestion. Second, by incorporating the relative importance of the media packets in the scheduling decisions at S-IMUX  218  at the sender, system  200  “cross-layer” optimizes media communication. In combination with the transmission over multiple TCP connections, this cross-layer optimization further improves the audio-visual quality of the media presentation at the receiver. 
         [0026]    The operation of system  200  in a communication session is described herein with reference to  FIGS. 3 and 4 , which show the operations of S-IMUX  218  and F-MUX  228 , respectively. At the beginning of the communication session, sender  210  establishes a plurality of TCP connections with receiver  220  (e.g., ‘N’ connections, where N is a positive integer and is a design parameter).  FIG. 3  shows the operation of S-IMUX  218  with the N connections already in place and available for use. As media packets are passed by scalable encoder  212  to S-IMUX  218  they are placed in an input buffer  310 . S-IMUX  218  then decides if and when to transmit the buffered packets (e.g., Pj-Pk), and over which of the N connections to transmit each of the packets. The decisions are made by S-IMUX  218 &#39;s Scheduling and Routing Unit (SRU)  320 , which makes the decisions based on the importance of the individual packets (e.g., relative to the scalability structure) and also on feedback received from receiver  220 &#39;s F-MUX  228  in the form of acknowledgement packets (e.g., ACKj-ACKk). SRU  320  includes suitable scheduling algorithms for this purpose. When a packet (e.g., Pj) is to be transmitted over a given connection (e.g., connection 1), it is placed in that connection&#39;s Output Buffer  330  until removed or replaced by SRU  320 . 
         [0027]    With continued reference to  FIG. 3 , assume that packets P j , . . . , P k , . . . , for k&gt;j, are provided by scalable encoder  212  to S-IMUX  218  and transmitted on Connection  1 , . . . , Connection  0 , . . . at times t j , . . . , t k , . . . , respectively, as decided or determined by SRU  320 . Receiver  210 &#39;s F-MUX  228  acknowledges the receipt of these packets via acknowledgement packets (ACK j -ACK k ) sent back to SRU  320 . The arrival times of the respective acknowledgements at the sender are denoted by t′ j , . . . , t′ k , . . . . 
         [0028]    Error control in SRU  320 &#39;s scheduling algorithm may be incorporated in the following manner. Let the current packet operated on by SRU  320  be P j . SRU  320  transmits packet P j  on Connection  1  at the time instance t j . SRU  320  then waits up to ‘T’ units of time to receive the corresponding acknowledgement on Connection  1 , where T is a design parameter. If an acknowledgement arrives by time t j +T, SRU  320  proceeds to the next packet in the input buffer. If, however, no such acknowledgement packet has arrived by time t j +T, SRU  320  flags Connection  1  as being unavailable at the moment (due to packet loss or congestion experienced thereon) and prepares for other packet scheduling steps. It is noted that TCP will continue trying to deliver this packet P j  on Connection  1  due to its property of reliable delivery. 
         [0029]    The next step in SRU  320 &#39;s packet scheduling procedure depends on the importance of packet P j . A “key video picture” or “key audio frame” (or parts thereof) is a picture or audio frame for which delivery is necessary in order to ensure an uninterrupted visual experience of the media presentation at the receiver. In scalable coding a key picture or key audio frame corresponds to the lowest temporal layer across all scalability dimensions provided by the encoder. In the following description, all such packets are referred to as key packets, without differentiating whether the encoded media is audio or video. 
         [0030]    If the unacknowledged packet P j  is not a key packet, then it is not retransmitted. S-IMUX  218  discards P j  and all subsequent packets received from scalable encoder  212  until a new key video picture or audio frame packet P k , for k&gt;j, is received for transmission. S-IMUX  218  then proceeds to transmit this new packet using the procedure described above for packet P j . 
         [0031]    If the unacknowledged packet P j  is a key packet, SRU  320  checks in a round-robin fashion if another connection (e.g., Connection  2 ) can be used to retransmit packet P j . SRU  320  may do this, for example, by verifying that the last packet sent on a particular connection (e.g., Connection  2 ) has been eventually acknowledged, i.e., it is no longer marked or flagged as unavailable. If that is the case, SRU  320  then transmits packet P j  on Connection  2 . SRU  320  will repeat the process of retransmitting packet P j  over other connections scanned in a round-robin fashion, until eventually the packet is acknowledged on one of the connections. When one such acknowledgement arrives, SRU  320  is done with packet P j  and can move on to transmitting another packet from the input buffer  310 . This other packet is not necessarily the packet immediately following P j  in input buffer  310 . 
         [0032]    When the receipt of key packet P j  is acknowledged after an initial failed transmission attempt, SRU  320  is in a congestion recovery mode. In order to minimize the amount of data to be transmitted, SRU  320  selects the next packet for transmission to be either the earliest key packet present in input buffer  310  or, if no such packet is yet available, it selects the latest packet P k , where k&gt;j. In this process, SRU  320  will skip over to the selected key packet in input buffer  310 , and discard (i.e., not transmit) all other in-between packets received from scalable encoder  212 . Transmission of the selected packet proceeds in the same manner as described herein. 
         [0033]    SRU  320 &#39;s scheduling algorithm is designed to allow the communication network to recover from the temporary congestion as detected by the missing acknowledgement ACK j  on Connection  1 . As SRU  320  sends no data until the next key picture (e.g., P k ) is due to be transmitted, SRU  320  in fact provides for faster congestion recovery of the communication network. Furthermore, by design, the intervening packets discarded by SRU  320  are not crucial for the continuous reconstruction of the media presentation at the receiver. It is expected that the temporary reduction in visual or audio quality of the presentation at the receiver due to non-receipt of the intervening packets is not dramatic, due to the scalable nature of the media encoding. 
         [0034]    It is noted that the scheduling algorithm of SRU  320  may continue to use a particular connection for subsequent transmissions of new packets, as long as the previous transmissions (on this same connection) are acknowledged in a timely manner (e.g., within the timer expiration limit T). While a connection is healthy (i.e., it has not timed out on a transmission), there is no reason to switch to any of the other N−1 TCP connections. Continued use of a healthy connection allows the other connections to remain open to potentially receive any pending acknowledgements for recent transmissions thereon, and thereby indicate recovery from congestion and/or packet loss that might have affected some of them recently. 
         [0035]    The detailed processing steps of SRU  320  are listed in TABLE I using pseudo-code. 
         [0036]    In TABLE I, n ε{0, 1, . . . , N−1} represents the connection number, P is the current packet, t denotes the current system time, and t 0  is a helper variable that stores time values. The flag ‘s’ is used to signal if packet skipping in input buffer  310  has to occur after an initial failed transmission attempt of a key packet (i.e., the first transmission of a packet timed-out). The flag is not necessary for non-key packets, as they are not retransmitted and the skipping can occur immediately. The function Free(n) is defined to return a 0 if connection ‘n’ is currently waiting for an acknowledgement packet and is thus unavailable for transmission, and 1 otherwise. Free(n) can be trivially implemented by associating a parameter ‘ack_state’ with each connection, which is set to 1 when a packet is transmitted, and reset when the corresponding acknowledgement is received. In such implementation, Free(n) simply returns the value of that flag for connection n. It is assumed that ACKs received at S-IMUX  218  are processed asynchronously to the processing steps shown below. 
         [0000]    
       
         
               
             
               
             
               
               
             
               
               
               
             
               
             
               
               
             
               
               
             
               
             
               
               
             
               
             
               
               
             
               
               
               
             
               
               
             
               
               
               
             
               
               
             
               
               
             
               
               
             
               
               
               
             
               
               
             
               
               
               
             
               
               
             
               
               
             
               
               
             
               
               
               
             
               
               
             
               
               
               
             
               
               
               
             
               
               
             
               
               
             
               
               
             
           
               
                 TABLE I 
               
               
                   
               
               
                 SRU 218 PROCESSING STEPS: 
               
               
                   
               
             
             
               
                   
               
             
          
           
               
                 (1) Initialize: 
               
             
          
           
               
                   
                 P := next packet from Input Buffer 
               
             
          
           
               
                   
                 s := 0 
                 /* reset skip flag */ 
               
             
          
           
               
                 (2) Get a free connection: 
               
             
          
           
               
                   
                 while (! Free(n) ) 
               
             
          
           
               
                   
                 n := (n+1) mod N 
               
             
          
           
               
                 (3) Transmit: 
               
             
          
           
               
                   
                 Transmit P on Connection n 
               
               
                   
                 t 0  := t 
               
             
          
           
               
                 (4) ACK or time out: 
               
             
          
           
               
                   
                 if (ACK received) 
               
             
          
           
               
                   
                 if (s == 1) 
                  /* check if we timed-out before, so we have 
               
             
          
           
               
                   
                  to skip */ 
               
             
          
           
               
                   
                 do 
                 /* skip to earliest key, or last Input Buffer 
               
             
          
           
               
                   
                 packet */ 
               
             
          
           
               
                   
                 P := next packet from Input Buffer 
               
             
          
           
               
                   
                 while (P != key packet) 
               
             
          
           
               
                   
                 else 
                  /* we didn&#39;t time-out before, no skip */ 
               
             
          
           
               
                   
                 P := next packet from Input Buffer 
               
             
          
           
               
                   
                 s := 0 
                 /* reset skip flag for the next packet */ 
               
             
          
           
               
                   
                 GOTO “(2) Get a free connection” 
               
             
          
           
               
                   
                 elsif (t &gt; t 0 +T) /* time-out occurred */ 
               
             
          
           
               
                   
                 If (P == key packet) 
               
             
          
           
               
                   
                 s := 1 
                  /* signal that we timed-out at least once */ 
               
             
          
           
               
                   
                 GOTO “(2) Get a free connection” 
               
             
          
           
               
                   
                 else 
                  /* not a key packet */ 
               
             
          
           
               
                   
                 do 
                 /* skip immediately to next key packet */ 
               
             
          
           
               
                   
                 P := next packet P from Input Buffer 
               
             
          
           
               
                   
                 while (P != key packet) 
               
               
                   
                 GOTO “(2) Get a free connection” 
               
             
          
           
               
                   
                 END 
               
               
                   
                   
               
             
          
         
       
     
         [0037]    The value for the time-out parameter T is preferably selected in consideration of the round-trip time (RTT) observed on the network path between sender  210  and receiver  220 . In particular, a judiciously selected T would not incur unnecessary retransmissions of media packets due to the late arrival of acknowledgements for the previous transmissions. At the same time, T should not unnecessarily delay retransmissions waiting for acknowledgements that will never materialize at the sender. Furthermore, the value selected for T must also account for the dynamics of the RTT over time and the related dispersion of its values. The processing steps listed in TABLE 1 may further include an upper limit on the number of retransmission attempts for a key frame, after which the connection is considered lost or not in service. This upper limit may be expressed by a second time-out parameter, T 2 , which may be set at a value several times that of parameter T. 
         [0038]    One approach to take into account all these requirements is to select T in the same way as TCP, where T is computed as mean(RTT)+α*std(RTT), where the multiplier α has the value 3 or 4. This quantity is dynamically updated as the values of the mean RTT and its standard deviation are (re)computed over time (i.e., online). To this end, the statistics of the RTT can be computed online by sender  210  based on the ACK packets or, if RTCP reports are available in system  200 , they can be obtained through their periodic exchange between senders and the receivers. 
         [0039]      FIG. 4  shows the operation of F-MUX  228  at receiver  220  corresponding to the operation of S-IMUX  218  with the N connections already in place and available for use ( FIG. 3 ). The N TCP connections (e.g., Connection  0 , . . . , Connection N−1) are terminated at the F-MUX Feedback and Combiner Unit (FCU)  410 . As soon as a packet P j  arrives on a connection n, a corresponding feedback packet ACK j  is transmitted back from FCU  410  to sender  210  on the same connection n. FCU  410  is also responsible for reassembling the different packets arriving on the multiple connections into a single packet stream created in F-MUX output buffer  430 , as well as for discarding duplicate packets that may be received due to the retransmissions performed by S-IMUX  218 &#39;s SRU  320 . FCU  410  is also connected to a set of N connection buffers  420  (‘Connection  0  Buffer’ through ‘Connection N−1 Buffer’), which are used by the FCU to temporarily store incoming packets for reordering and to also know which packets have already been received so that duplicates are eliminated. 
         [0040]    The proper ordering of incoming packets for the single packet stream created in F-MUX output buffer  430  is dependent on the particular scalability structure used in system  200 . As an illustrative example, assume that scalable encoder  212  ( FIG. 2 ) is a scalable video encoder that operates with two spatial layers and three temporal layers, as described in International Patent Application PCT/US06/028365.  FIG. 5  shows an exemplary picture coding structure  500  for such an encoder. The prediction paths in structure  500  that may be used by encoder  212  are designated by arrows. The key packets for such an encoder are those that carry L 0  data, i.e., the lowest spatial and temporal layer. The systems and methods of the present invention are designed to ensure the timely delivery of this data, as it is crucial for decoding of the data of all other layers (L 1 -L 2 , S 0 -S 2 ). 
         [0041]    In this example, FCU  410  will have to create an output packet stream in the output buffer  430  so that lower layers precede higher layers for the same temporal instance, while maintaining-proper temporal ordering of pictures (in coding order). As an example, consider that the four pictures (e.g., (L 0 , S 0 ) . . . (L 2 , S 2 )) shown in  FIG. 5  are received in different packets across a set of N connections. Due to the design of S-DMUX  218  (application-level retransmissions over TCP connections that ultimately ensure reliable delivery), it is theoretically possible to have multiple copies of a packet arriving at FCU  410  as well as out of order arrivals. In response to such situations, FCU  410  then places arriving packets in the corresponding connection buffers  420 , and at the same time it continuously tries to assemble the output buffer  430  stream by including, in sequence, packets for L 0 , S 0 , the first L 2  and S 2 , L 1  and S 1 , and finally the second L 2  and S 2  (assuming all layers are to be received at the FCU). In doing so, FCU  410  examines all connection buffers  420  to find the appropriate packet at each step in sequence. When such a packet is found, it is removed from the buffer where it is located. While searching for the correct packet, FCU  410  can also remove duplicate packets that are no longer needed using the timing or picture ordering information that all standard codecs embed in their packetized data. 
         [0042]    The embodiments of the invention as described above assumes that the internal TCP control parameters are not available to the application level. In other words, the TCP/IP components of the sender and receiver are assumed to be “black boxes,” and accessible only through their standard interfaces (e.g., sockets). When access to TCP source code is available to the designer, it may be possible to utilize TCP&#39;s acknowledgement status information and to thereby avoid transmitting an application-level acknowledgment packet from the receiver to the sender, in accordance with the present invention. The bit rate savings, however, may not be very significant, especially in a two-way communication system where large amounts of media data flow in both directions. 
         [0043]    It will be understood that in accordance with the present invention, the transmission techniques described herein may be implemented using any suitable combination of hardware and software. The software (i.e., instructions) for implementing and operating the aforementioned rate estimation and control techniques can be provided on computer-readable media, which can include, without limitation, firmware, memory, storage devices, microcontrollers, microprocessors, integrated circuits, ASICs, on-line downloadable media, and other available media.