Patent Publication Number: US-2007097987-A1

Title: Feedback provision using general nack report blocks and loss rle report blocks

Description:
FIELD OF THE INVENTION  
      The invention relates to a method for providing RTCP feedback messages for data packets of at least one streaming session from a client to a streaming server, wherein the at least one streaming session is provided using a RTP protocol and the RTCP protocol. Further, a RTCP bandwidth, being a fraction of the available streaming session bandwidth, is allocated to the RTCP feedback messages. Moreover the invention further relates to a client in a mobile communication system performing the method.  
     BACKGROUND ART  
      The 3GPP (3 rd  Generation Partnership Project) adopts IETF (Internet Engineering Task Force) standardized protocols like RTP, UDP, IP for the transport and packet-switch codecs like AMR (Adaptive Multi-Rate) and H.264 (MPEG 4 part 10) for encoding media. The 3GPP Packet Switched Streaming Services (see “Universal Mobile Telecommunications System (UMTS); Transparent end-to-end streaming service; Protocols and codecs”, 3GPP TS 26.234 version 6.1.0, September 2004, available at http://www.3gpp.org) use the RTP/UDP protocol stack to stream audio/video/text media.  
      RTP is a Real-Time Transport Protocol (see Schulzrinne et al., “RTP: A Transport Protocol for Real-Time Applications”, RFC 3550, July 2003, all IETF RFCs and Internet Drafts available at http://www.ietf.org) which is mainly used for real-time or near real-time communication, i.e. communication with relaxed delay constraints. It provides information on the timing of the media it carries and also allows re-ordering and re-assembling at the receiver.  
      An integrated part of the protocol is RTCP (Real Time Control Protocol) which provides minimal reception information and loose group membership. RTP is generally used together with the RTP/AVP profile (see Schulzrinne et al., “RTP Profile for Audio and Video Conferences with Minimal Control”, RFC 3551, July 2003) which defines the use of the RTP header fields and mapping tables for payload types, besides simple RTCP feedback timing rules. The RTCP protocol is allocated a fraction of the available RTP bandwidth in order to compete with TCP/IP based traffic in a fair manner.  
      UDP (Postel, “User Datagram Protocol”, RFC 768, August 1980) is the User Datagram Protocol used to transport RTP packets. UDP is commonly used when an unreliable communication is appropriate for the given media, as is the case for streaming applications. The protocol stack RTP/UDP is used because the timing constraints of the media don&#39;t allow reliable communication, e.g. by using TCP (Transmission Control Protocol).  
      In RTP, packetization schemes (payload formats) for existing media formats (codecs) are specified in the Internet Engineering Task Force AudioNideo Transport Working Group (IETF AVT WG). There is, for example, a payload format for AMR encoded speech data, and another one for H.264 video. A RTP payload format defined in Rey et al., “RTP Retransmission Format”, Internet Draft, IETF AVT Workgroup, August 2003, also allows for retransmission of RTP packets in this framework. This may especially useful for 3GPP PSS streaming services.  
      As specified in by Rey et al. in “RTP Retransmission Payload Format” the mechanism for issuing multiple retransmissions relies on the client issuing the request only when needed. The client must use the feedback possibilities specified in Ott et al., “Extended RTP Profile for RTCP-based Feedback (RTP/AVPF)”, Internet Draft, IETF AVT Workgroup, August 2004. Other reporting blocks, such as those defined by Friedman et al. in “RTP Control Protocol Extended Reports (RTCP XR)” (see RFC 3611, November 2003) may also be used.  
      The frequency of reporting may be calculated using the standard RTCP report interval calculation in RFC 3550 (see section 6.3 and Annex). The coverage of the receiver reports, i.e. how many packets are reported, is not specified. The client uses General NACK (Negative ACKnowledgement) feedback messages as specified in the RTP/AVPF profile (draft-ietf-avt-rtcp-feedback-11.txt, can be obtained at http://www.ietf.org/internet-drafts/). Note that the client cannot acknowledge packets using the RTP/AVPF profile since there are no ACK messages specified there. Thus, no provision for comprehensive reporting e.g. by using RLE reports is made.  
      Furthermore, the Loss RLE Reports as defined in RFC 3611 are enhanced reporting blocks, which are compressible and combine ACKs and NACKs per default, i.e. allow to report on lost and received packets. These reports have been specified for network maintenance purposes, charging, multicast tree inference and statistics gathering.  
      A third document that deals with Loss RLE Reports and unicast RTCP reporting is the 3GPP Packet Switched Streaming Services (PSS) framework. According to this document it is recommended that streaming clients implement the usage of Loss RLE Report blocks and report in a redundant manner over past RTCP intervals. Further, it is noted that these reports should not be used to trigger retransmissions according to the 3GPP PSS framework.  
      Redundant reporting in the context of the 3GPP PSS framework is understood as enabling a minimum number of reports on lost data packets prior to or after their processing. There is, however, no specification as to how this mechanism should be implemented at the client and how this mechanism should be used together with RTP retransmissions requests. Redundant reporting on lost packets is desirable since feedback, i.e. RTCP messages, is commonly transmitted using unacknowledged and thus unreliable transport protocols like UDP. Therefore, the feedback provided by the receivers of a session may be lost especially when considering the provision of services via wireless access networks, as for example UMTS.  
      As indicated above, the PSS framework requires that Loss RLE Report blocks (RFC 3611) are used for charging and network monitoring purposes only, while General NACK messages (RTP/AVPF profile) are to be used to actually trigger retransmissions of the lost data packets. Thus, the PSS framework requires that the RTCP feedback of a transport session comprises General NACKs and Loss RLE Report blocks in order to allow triggering retransmissions and simultaneously allow (e.g.) charging and network monitoring.  
      The multiple retransmission algorithms described by Rey et al. uses (or might make use of) message formats defined in the RTP/AVPF draft by Ott et al., i.e. General NACKs, to trigger packet retransmission.  
      If both received and not received packets shall be reported, as recommended by the 3GPP PSS framework, this means that, these report blocks can build up to a considerable overhead, such that the reserved bandwidth allocated to RTCP might not be sufficient to report over the required amount of time to guarantee continuous play-out and/or redundancy of the reports as required. Further, in order to enable charging and network monitoring services in parallel to triggering retransmission, the feedback of the clients receiving a PSS session actually needs to include redundant report blocks (Loss RLE Report blocks), which also causes overhead and impacts the reporting frequency and thereby the level of redundant reporting that may be provided.  
     SUMMARY OF THE INVENTION  
      The object of the invention is therefore to specify method for optimizing RTCP feedback in a unicast sessions.  
      The object of the invention is solved by the subject-matters of the independent claims.  
      Preferred embodiments are subject matter to the dependent claims.  
      One embodiment of the invention relates to a method for providing RTCP feedback messages for data packets of at least one streaming session from a client to a streaming server, wherein the at least one streaming session is provided using a RTP protocol and the RTCP protocol. According to this method, the client may determine first, if a maximum number of retransmissions for a data packet that is providable within session constraints of the streaming session is larger than a minimum number of retransmissions to be enabled by the client. If so, a number of General NACK report blocks may be added to a RTCP feedback packet for requesting the retransmission of lost data packets. The added number of General NACK report blocks enables the minimum number of retransmissions of the lost data packets within a client buffering time. Next, the client may determine the maximum RTCP feedback message size that is allowable in view of the session constraints, and may add a number of Loss RLE report blocks to the RTCP feedback packet for reporting on lost data packets, such that size of the resulting RTCP feedback packet does not exceed the maximum RTCP feedback message size. Further, the client may transmit the RTCP feedback packet as an RTCP feedback message to the streaming server.  
      Another embodiment of the invention relates to a method for providing RTCP feedback messages for data packets of at least one streaming session from a client to a streaming server. In this embodiment, at least one streaming session is provided using a RTP protocol and the RTCP protocol, and a RTCP bandwidth, being a fraction of the available streaming session bandwidth, is allocated to the RTCP feedback messages.  
      The client may receive session description information being indicative of the RTCP bandwidth for the RTCP feedback messages, a minimum number of retransmissions to be enabled within the client buffering time, a minimum reporting redundancy for lost data packets of the at least one streaming session and the client buffering time. Further, the client may determine the maximum number of retransmissions for a data packet that is providable according to based on the client buffering time and a RTCP report interval. The RTCP report interval may thereby depend on an average RTCP feedback message size and the RTCP bandwidth allocated to the client.  
      The client may further determine whether the maximum number of retransmissions is larger than or equal to the minimum number of retransmissions, and if so, it may add a number of General NACK report blocks to a RTCP feedback packet for requesting the retransmission of lost data packets. This number of General NACK report blocks added to the RTCP feedback packet may enable the minimum number of retransmissions of the lost data packets within the client buffering time.  
      The client may also determine the maximum RTCP feedback message size that is allowable under consideration of the allocated RTCP bandwidth, the client buffering time and the minimum number of retransmissions and may determine a resulting average RTCP feedback message size based on the average size of previously sent RTCP feedback message and the size of the RTCP packet comprising the number of General NACK report blocks.  
      Further, the client may add a number of Loss RLE report blocks to the RTCP feedback packet for reporting on the lost data packets, if the size of the resulting RTCP feedback packet does not exceed the maximum RTCP feedback message size, and may transmit the RTCP feedback packet as an RTCP feedback message from the client to the streaming server.  
      In another embodiment of the invention the Loss RLE report blocks are added to the RTCP feedback packet for charging or network monitoring purposes.  
      In a further embodiment the number of Loss RLE report blocks is run-length encoded prior to determining, if the size of the resulting RTCP feedback packet including the run-length encoded Loss RLE report blocks does not exceed the maximum RTCP feedback message size. By performing a run-length encoding of Loss RLE report blocks, their size may be reduced while the reporting redundancy that may be provided with the respective number of report blocks remains constant.  
      In another embodiment, the client may decide to reduce the minimum reporting redundancy configured by the session description information by the client. Different possibilities to reduce the reporting redundancy on packets lost may be foreseen.  
      For example in one variation of this embodiment the client may determine the uplink quality of service that is provided for packets of the streaming session based on this measured or determined quality of service (QoS), the minimum reporting redundancy is reduced, for example if the determined quality of service is above a predetermined threshold level.  
      Another variation foresees that the reporting redundancy is reduced by the client by reducing the size of the number of Loss RLE report blocks.  
      For example, in case the data of the streaming session comprises a basic data layer providing a basic streaming media quality and at least one enhancement layer enhancing the basic streaming media quality, the size of the number of Loss RLE report blocks may be reduced by including only information related to data packets that comprise data of the basic data layer to within the Loss RLE report blocks.  
      When for example the data of the at least one streaming session is an MPEG stream and the basic data layer comprises I-frames and the at least one enhancement layer comprises P-frames and/or B-frames. In this example, the client may ensure that I-frames are reported on while P-frames and/or B-frames which have less impact on the clients perceived QoS of the stream may be only reported if possible within the session constraints set by the session description information.  
      In this respect, the client may determine whether a lost data packet comprises data of the basic data layer based on data packets already received at the client.  
      A further possibility to reduce the size of the number of Loss RLE report blocks according to a further variation of the embodiment is to apply thinning to the Loss RLE report block(s).  
      Moreover, another variation of the embodiment foresee that the size of the number of Loss RLE report blocks is reduced by reducing the number of Loss RLE report blocks being reported in the RTCP feedback packet.  
      Ideally, in another embodiment of the invention the number of Loss RLE report blocks to be added to the RTCP feedback packet enables the minimum reporting redundancy of the lost packets.  
      Further, another embodiment of the invention foresees that a MIME type parameter of the streamed session indicates the packet rate. This parameter may for example useful to estimate the average number of packets of a streaming session that will have to be reported within a RTCP report interval.  
      In a further embodiment of the invention the client may determine which packets of the streaming session are lost based on sequence numbers of data packets received at the client.  
      Another embodiment of the invention relates to a client in a mobile communication system transmitting RTCP feedback messages for data packets of at least one streaming session from a client to a streaming server. Again the at least one streaming session is provided using a RTP protocol and the RTCP protocol, and a RTCP bandwidth, being a fraction of the available streaming session bandwidth, is allocated to the RTCP feedback messages. The client according to this embodiment of the invention may comprise a receiver for receiving session description information, wherein the session description information is indicative of the RTCP bandwidth for the RTCP feedback messages, a minimum number of retransmissions to be enabled within the client buffering time, a minimum reporting redundancy for lost data packets of the at least one streaming session and the client buffering time, and processing means for determining the maximum number of retransmissions for a data packet that is providable according to based on the client buffering time and a RTCP report interval, wherein the RTCP report interval depends on an average RTCP feedback message size and the RTCP bandwidth allocated to the client, and for determining whether the maximum number of retransmissions is larger than or equal to the minimum number of retransmissions.  
      The processing means may be further adapted to add a number of General NACK report blocks to a RTCP feedback packet for requesting the retransmission of lost data packets, wherein the number of General NACK report blocks added to the RTCP feedback packet enables the minimum number of retransmissions of the lost data packets within the client buffering time, and to determine the maximum RTCP feedback message size that is allowable under consideration of the allocated RTCP bandwidth, the client buffering time and the minimum number of retransmissions, if the maximum number of retransmissions is larger than or equal to the minimum number of retransmissions.  
      The processing means may also be adapted to determine an resulting average RTCP feedback message size based on the average size of previously sent RTCP feedback message and the size of the RTCP packet comprising the number of General NACK report blocks, and to add a number of Loss RLE report blocks to the RTCP feedback packet for reporting on the lost data packets, if the size of the resulting RTCP feedback packet does not exceed the maximum RTCP feedback message size.  
      The client may further comprise a transmitter for transmitting the RTCP feedback packet as an RTCP feedback message to the streaming server.  
      Another embodiment relates to the client, further comprising means adapted to perform the method according to one of the various embodiment described above and variations thereof.  
      Moreover another embodiment of the invention relates to a computer readable medium for storing instruction that, when executed by a processor of a client in a mobile communication system, cause the client to provide RTCP feedback messages for data packets of at least one streaming session from a client to a streaming server. Again, the at least one streaming session is provided using a RTP protocol and the RTCP protocol, and a RTCP bandwidth, being a fraction of the available streaming session bandwidth, is allocated to the RTCP feedback messages.  
      The client may be caused to provide RTCP feedback messages for data packets of at least one streaming session from a client to a streaming server by receiving session description information at the client, wherein the session description information is indicative of the RTCP bandwidth for the RTCP feedback messages, a minimum number of retransmissions to be enabled within the client buffering time, a minimum reporting redundancy for lost data packets of the at least one streaming session and the client buffering time, by determining the maximum number of retransmissions for a data packet based on the client buffering time and a RTCP report interval, wherein the RTCP report interval depends on an average RTCP feedback message size and the RTCP bandwidth allocated to the client, by determining whether the maximum number of retransmissions is larger than or equal to the minimum number of retransmissions.  
      If the latter is the case, the instructions may cause the client to adding a number of General NACK report blocks to a RTCP feedback packet for requesting the retransmission of lost data packets, wherein the number of General NACK report blocks added to the RTCP feedback packet enables the minimum number of retransmissions of the lost data packets within the client buffering time, to determine the maximum RTCP feedback message size that is allowable under consideration of the allocated RTCP bandwidth, the client buffering time and the minimum number of retransmissions, to determining an resulting average RTCP feedback message size based on the average size of previously sent RTCP feedback message and the size of the RTCP packet comprising the number of General NACK report blocks, to add a number of Loss RLE report blocks to the RTCP feedback packet for reporting on the lost data packets, if the size of the resulting RTCP feedback packet does not exceed the maximum RTCP feedback message size, and to transmitting the RTCP feedback packet as an RTCP feedback message from the client to the streaming server.  
      The computer readable medium according to a further embodiment of the invention, further stores instructions that, when executed by the processor of the client, cause the client to perform the method according to one of the various embodiments of the invention described above an variations thereof.  
      Another embodiment of the invention provides a system comprising a streaming server providing at least streaming session to a client, wherein the at least one streaming session is provided using a RTP protocol and the RTCP protocol, and a client according to one of the embodiments of the invention above. 
    
    
     BRIEF DESCRIPTION OF THE FIGURES  
      In the following the invention is described in more detail in reference to the attached figures and drawings. Similar or corresponding details in the figures are marked with the same reference numerals.  
       FIG. 1  shows an overview of a streaming environment according to an embodiment of the invention,  
       FIG. 2  shows the relationship between the client buffering time, the RTCP report period and a minimum time period required to provide a target level of retransmissions according to an embodiment of the invention, and  
       FIG. 3  shows the timeline of a number of consecutive retransmission requests and retransmissions in a streaming environment according to an exemplary embodiment of the invention,  
       FIG. 4  shows the initial steps of a method for providing RTCP feedback within a streaming session according to an embodiment of the invention, and  
       FIGS. 5 &amp; 6  show further steps of the method for providing RTCP feedback according to different embodiments of the invention, wherein the illustrated steps are performed after those shown in  FIG. 4 . 
    
    
     DETAILED DESCRIPTION  
       FIG. 1  shows an overview of a streaming environment according to an embodiment of the invention. A streaming server  100  may deliver a streaming service in form of one or multiple sessions via a wireless access network to the client  101  or mobile terminal. In the example a UMTS network comprising a core network  103  (CN) and a UMTS radio access network  104  (UTRAN) provides the streamed packet switched service, which may be operated according to the requirements as defined in 3GPP TS 26.234. To connect the core network  103  to a packet switched network, e.g. the Internet, the core network may comprise a gateway GPRS support node  105  (GGSN) and a serving GPRS support node  106  (SGSN). The core network&#39;s components may be connected to the UTRAN  104  comprising at least one radio access controller  107  (RNC) and at least one Node B connected to a RNC. The data of the streamed media may be provided to a mobile client  101  via a wireless link.  
      One aspect of the invention is to efficiently use RLE report blocks for loss reporting in different RTP retransmission implementations. Such implementation typically differ in the functionalities that server and client carry out, the information both use to schedule/request retransmissions and the knowledge that they have about the network conditions (congestion, packet losses, link characteristics). The 3GPP PSS streaming services define session description attributes, which may or may not be used to make this information available between communicating peers.  
      For all RTP retransmission implementations the same initial scenario and requirements may be assumed: The client should perform redundant reporting as RTCP transport is unreliable. Further, the client should report at least over the period between the current report and the last report, i.e. guarantee minimum receiver reporting.  
      The report frequency should be sufficiently high to allow for timely retransmission, i.e. to guarantee a minimum number of retransmissions for each RTP packet. Accordingly, the client may limit its report frequency, report block size and redundant reporting to comply with the RTCP bandwidth signaled by the server.  
      As some of these requirements are competing and it might not be possible to fulfill them simultaneously in every scenario, the client may optimize the level of redundant reporting, the reporting frequency and the number of reported RTP packetst for the given conditions.  
      For example (and assuming that the RTCP bandwidth share and the client buffering time allows for redundant reporting) this could mean that in an simple implementation the client will request retransmission for and report about all lost packets between the current RTCP report and the last RTCP report using General NACK report blocks and Loss RLE report blocks, respectively, set the level of redundant reporting on lost and received packets to a reasonable value and keep the RTCP report frequency as high as possible to allow for a maximum of timely retransmissions. It should be noted that RTP packets on the downlink (server to client) may be lost multiple times and may therefore require multiple retransmissions.  
      If this simple implementation should not comply with the requirement to limit the report frequency, report block size and redundant reporting to comply with the signaled RTCP bandwidth, the client may first try to reduce the Loss RLE Report block size by run length encoding.  
      A Loss RLE report block permits detailed reporting upon individual packet receipt and loss events. Such reports can be used, for example, for multicast inference of network characteristics (MINC). Since a Boolean trace of lost and received RTP packets is potentially lengthy, this block type may permit the trace to be compressed through run length encoding. When using run length encoding, the trace to be encoded is analyzed and “runs” of identical events may be summarized to reduce the overall size of the Loss RLE report block.  
      Next, the client may reduce the number of reported RTP packets per Loss RLE report block. Please note, that the reduced number of packets that are reported by a single report block will result in a reduced level of redundant reporting.  
      In order to reduce the number of packets that is reported on by a Loss RLE report block, loss event reports can be systematically dropped from the trace in a mechanism called thinning. The thinning mechanism may be implemented by choosing a thinning value, T, which may be used to select a subset of packets within the sequence number space, e.g. those packets with sequence numbers that are multiples of 2T. Packet reception and loss reports apply only to those packets. T may e.g. vary between 0 and 15. If T is zero, then every packet in the sequence number space is reported upon. If T is fifteen, then one in every 32,768 packets is reported upon.  
      Suppose that the trace just described begins at sequence number 13,821. The last sequence number in the trace is 13,865. If the trace were to be thinned with a thinning value of T=2, then every fourth sequence numbers would be reported upon: 13,824, 13,828, 13,832, 13,836, 13,840, 13,844, 13,848, 13,852, 13,856, 13,860, 13,864.  
      Another option to reduce the number of packets reported on is to report only on the packets that fall in the latest packets.  
      Finally if all these measures are not helpful and if there is a hard requirement for a minimum number of retransmission requests (General NACK report block(s)) prior to expiry of the client buffering time, the client may restart the session with enhanced RTCP bandwidth to comply with these required minimum number of retransmissions and/or may enhance the client buffering time (see below for more details).  
      For the further discussion of the ideas underlying the invention, the relations between RTCP bandwidth, client buffering time, RTCP report interval, etc. will be discussed with reference to  FIG. 3 .  FIG. 3  shows an overview of a streaming environment according to an embodiment of the invention. In the Figure the timeline of a number of consecutive retransmission requests and retransmissions is shown. The initial transmission of the RTP packet (SN=0) as well as the first and second retransmission thereof is assumed to be lost.  
      The sender (e.g. streaming server) provides a RTP data packet to the receiver (client). The parameters contributing to the delay T 1  are the one-way-delay from sender to receiver on through network(s), i.e. the physical delay, the transmission delay on the air interface and the delay caused by interarrival jitter caused by network queues, reordering, etc.: 
 
 T   1 =physical delay+transmission delay+interarrival delay jitter  (1) 
 
 or more precise:  
                     T   1     =       ⁢       physical   ⁢           ⁢   delay     +       average   ⁢           ⁢   packet   ⁢             ⁢             ⁢   size       available   ⁢           ⁢   downlink   ⁢             ⁢             ⁢   bitrate       +                     ⁢     interarrival   ⁢           ⁢   delay   ⁢             ⁢             ⁢   jitter                   (   2   )             
 
      Assuming that the initial transmission of the packet is lost, T 2  specifies the time interval between the theoretical arrival of the packet and the point in time at which the receiver provides feedback indicating the packet loss to the sender.  
      Thereby T 2  is composed of the time required to detect the packet loss (packet loss detection time) at the client and the average time to the next feedback report from the client: 
 
 T   2 =packet loss detection time+average time to next feedback report  (3) 
 
      The packet loss detection time depends on whether single packets are occasionally lost (single-ton losses) or whether the packet losses occur in bursts (bursty losses). In the first case, the packet loss detection time should be equal to the interarrival delay jitter experienced at the receiver. In the second case an additional guard time must be added to the buffering time for the determination of the actual value. This guard time may be defined as the average loss burst and reflects the fact that bursty losses take a longer time to be detected.  
      Commonly loss events are uniformly distributed such that the average time to the next feedback report is commonly one half of the RTCP report interval: 
 
average time to next feedback report=½ ·RTCP  report interval  (4) 
 
      The delay of the feedback report T 3  may be essentially calculated in a similar manner as the delay T 1 . 
 
 T   3 =physical delay+transmission delay+interarrival delay jitter  (5) 
 
 or more precise:  
                     T   3     =       ⁢       physical   ⁢           ⁢   delay     +       average   ⁢           ⁢   packet   ⁢             ⁢             ⁢   size       available   ⁢           ⁢   uplink   ⁢             ⁢             ⁢   bitrate       +                     ⁢     interarrival   ⁢           ⁢   delay   ⁢             ⁢             ⁢   jitter                   (   6   )             
 
      Finally, the processing time of the retransmission at the sender is denoted T 4  and comprises the time needed by the sender to decide which packets are to be retransmitted (feedback processing time) and the queuing time at the sender, i.e. the time the packets to be retransmitted stay in the (re)transmission queue of the sender before being actually transmitted: 
 
 T   4 =feedback processing time+queuing time  (7) 
 
      It should be noted that the T 1 +T 2 +T 3  corresponds to the roundtrip time (RTT): 
 
 RTT=T   1   +T   2   +T   3   (8) 
 
      The RTT is typically smaller than the RTCP report interval. Otherwise the reporting would be excessive, since the receiver would send feedback without awaiting the retransmission of packets first. Furthermore the sum of the RTT plus the feedback processing time likewise typically falls within the RTCP report interval since the server is usually subject to guarantee a defined retransmission bandwidth and thus this time does not increase much.  
      Another important “variable” of a RTP session having impacting the reporting redundancy that can be provided, is the client buffering time already mentioned above. The client buffering time may be generally defined as the time span a particular data packet is buffered in a memory of the receiver before being “output” to the user. For example, for a video streaming session, the client buffering time may be understood as the time span of storing a data packet in the receiver&#39;s memory (buffer) before displaying the content of the data packet to the user.  
      The number of retransmissions that may be provided in a RTP session before a packet is provided to the user may be specified as:  
                     no   .           ⁢   of             possible           retransmissions         =     Integer   (               client   ⁢           ⁢   buffering   ⁢           ⁢   time     -               (       T   1     +     T   2     +     T   3     +     T   4       )             RTCP   ⁢           ⁢   report   ⁢           ⁢   interval       )             (   9   )             
 
      The “Integer( )” function extracts the integer value from the result of the division. The RTCP report interval (for uplink transmissions) is defined by:  
                 RTCP   ⁢           ⁢   report   ⁢           ⁢   interval     =       average   ⁢           ⁢   RTCP   ⁢           ⁢   packet   ⁢             ⁢             ⁢   size       available   ⁢           ⁢   uplink   ⁢             ⁢             ⁢   RTCP   ⁢           ⁢   banwidth         ⁢     
     ⁢   wherein           (   10   )                             average   ⁢           ⁢   RTCP               packet   ⁢           ⁢   size           =       ⁢           1   /   16     ·   current     ⁢           ⁢   RTCP   ⁢           ⁢   packet   ⁢           ⁢   size     +                     ⁢         15   /   16     ·   average     ⁢           ⁢   RTCP   ⁢           ⁢   packet                     ⁢   size                 (   11   )             
 
      In this equation the current RTCP packet size equals the size of the next RTCP feedback to be sent.  
      One simplification of the calculations above may be that only non-bursty, i.e. singleton losses of packets are considered in the definition of T 2 , i.e. 
 
T 2 =interarrival delay jitter  (12) 
 
      Since T 1 +T 2 +T 3 +T 4  may not be always available at the client, a conservative approximation can be used for this sum may be used as indicated in  FIG. 3  (note that a minimum number of retransmissions of 3 is assumed for exemplary purposes). E.g. the RTCP report interval value may represent a rough estimate for this sum and a conservative buffering time client_buffering time′ which may be defined: 
 
client_buffering_time′=client_buffering_time−tcp_report_interval 
 
      The conservative client buffering time client_buffering time′ can also be understood as the buffering time to ensure the minimum number of required retransmissions min_no_rtcs. This is of course only valid as long as the value T 1 +T 2 +T 3 +t 4  is less than the RTCP report interval, which is typically the case since otherwise the client would not await a possible retransmission before sending the feedback. This shall be ensured by the server or content creator responsible for compiling the session description information using SDP or similar protocols.  
      According to one embodiment of the invention the calculation of the number of possible retransmissions (see Equation 9) may be simplified to:  
                           no   .           ⁢   of             possible           retransmissions         =       ⁢       Integer   ⁡     (               client   ⁢           ⁢   buffering   ⁢           ⁢   time     -               (       T   1     +     T   2     +     T   3     +     T   4       )             RTCP   ⁢           ⁢   report   ⁢           ⁢   interval       )       =                   ≈       ⁢     Integer   ⁡     (               client   ⁢           ⁢   buffering   ⁢           ⁢   time     -               RTCP   ⁢           ⁢   report   ⁢           ⁢   interval             RTCP   ⁢           ⁢   report   ⁢           ⁢   interval       )         =               =       ⁢     Integer   (       client   ⁢           ⁢   buffering   ⁢           ⁢     time   ′         RTCP   ⁢           ⁢   report   ⁢           ⁢   interval       )                   (   13   )             
 
      Of course also the more precise calculation of equation 9 may be used. Equation 13, may however be easier to implement. In the initial phase of a PSS streaming service the client requests a description on details of the session offered from a certain server. In response to the request, the server sends the session description. Inter alia, the following parameters are made known to the client by the session description: 
          MIME Type of the codec used for the media session: e.g. H263, AMR, MPEG4, etc.     RTCP bandwidth share, client_rtcp_bandwidth (bits per second), for the receiver reports. The default bandwidth share for unicast streaming applications such as those targeted by the PSS specification is 50% of the total RTCP bandwidth for each, sender and receiver. Another values may be explicitly specified using the provisions of RFC 3556. The total RTCP bandwidth is commonly 5% of the total session bandwidth.     Optionally MIME Type parameters that described the codes. These parameters may be used to calculate other values of the stream such as the packet rate (number of packets per second).     Optionally the maximum RTCP packet size that a client may use. Currently it is possible within the PSS framework to limit the RTP (not RTCP) packet size of a server. Similar limitations are possible for RTCP report packets in the future.        

      Further, according to an embodiment of the invention the following additional parameters are known to the client: 
          The minimum number of retransmissions, min_no_rtxs, recommended by the server. The server may propose that the client guarantees a minimum number of retransmissions for each packet. The reason for this may be that the service provider is aware of the packet loss on the link. As indicated above, retransmissions in PSS streaming services need to be triggered by General NACK report blocks in the feedback.     The reporting redundancy, report_redundancy, (also) recommended by the server. This quantity indicates how many times a packet is reported using the Loss RLE report blocks. For service providers it is important to know the achieved performance of the link (which packets arrived or not) and how many retransmissions are needed at maximum to provide the packets to the clients. This is also important for charging, where clients are only charged for the individually perceived service quality.     The client buffering time, client_buffering_time (measured in seconds) that may be derived from the session parameters as described in the following paragraphs.        

      Considering the client buffering time there may be two possible scenarios:  
      The first possibility is that, the client does not support bit-rate adaptation as defined in the 3GPP PSS framework.  
      In this case the client_buffering_time, may be chosen equal to the initial buffering time, i.e. the time after which the client starts reading out the packets of the buffer at session start.  
      The second possibility is that the client supports bit rate adaptation as defined by the 3GPP PSS framework. In this latter case, the following parameters are available to the client: 
          The buffer size, buffer size, which is signaled in the “3GPP-Adaptation” header. This parameter should correspond to a reception and de-jittering buffer that has this given amount of space for complete RTP packets including the RTP header. The specified buffer size may also include any pre-decoder buffer space used for this media. This parameter thus corresponds to a maximum size of session data (RTP packets) the client may have available at any given moment of the session. Commonly, this parameter is not used for determining a value for the client buffering time.     The target protection time, target_protection_time, signaled to the client in the “3GPP-Adaptation” header (“target-time” parameter). This parameter denotes the targeted minimum buffer level or, in other words, the desired amount of playback time at the client (in milliseconds) to guarantee interrupt-free playback and allow the server to adjust the transmission rate, if needed. According to one embodiment of the invention, this parameter is used as the client buffering time, client_buffering_time.        

      Once the client has received the session description information the client may calculate whether the signaled RTCP bandwidth share (client_rtcp_bandwidth) and intended client_buffering_time for the given media stream is chosen such that reporting on packet losses is possible. Additionally it may further calculate, if the desired report redundancy as indicated by the report_redundancy parameter and the requested minimum number of retransmissions indicated by the min_no_rtxs parameter may be provided given the RTCP bandwidth share and client buffering time.  
      It should be recalled that the reporting redundancy refers to the provision network monitoring, charging, etc. issues (Loss RLE Report blocks), while the minimum number of retransmissions to be provided is based on the feedback of General NACKs, as outlined previously.  
      According to one embodiment it is assumed that the client&#39;s buffer sets a timer to client_buffering_time seconds for each packet. If a packet is not received within said that time span, the timer expires and the packet cannot be longer requested for retransmission (i.e. cannot be included in a further General NACK message).  
      Further, it may also be assumed that past RTCP reports (containing the previously transmitted General NACK report blocks and Loss RLE report blocks) are saved in a send buffer at the client. The size of this buffer may for example be set corresponding to the server_buffering_time (in seconds). Moreover, this buffer should big enough to store the needed RTCP packets sent, as needed by the method described below.  
      The following passages describe an exemplary algorithm according to an embodiment of the invention which is complementary to the standard RTP algorithms as described in RFC 3550, Annex A. In contrast to the standard RTP algorithms, the invention according to this algorithm allows considering the parameters min_no_rtxs and report_redundancy to configure a PSS conform session. The algorithm helps the client find out what is the size of the General NACK and Loss RLE Report needed to: 
          enable the maximum number of packet retransmissions for every packet or enforce a minimum number of retransmissions, min_no_rtxs, within the client buffering time     provide report redundancy as set by the report_redundancy parameter (if possible),     while complying with the given RTCP bandwidth for receiver reports and client buffering capabilities.        

      The following exemplary steps may be executed by the client for every RTCP receiver report sent by the client, starting with the first RTCP packet.  
      Step 1. After an initial waiting period of 1 second as per RTP/AVPF, the client may request packet retransmissions in a new General NACK message. From an implementation viewpoint, this step is a look-up operation since at every client implementing retransmission according to the Internet Draft of Rey et al. mentioned above should keep a list of pending requests.  
      An exemplary list may consist of two columns: one with the sequence number SN and other with a flag indicating whether the packet with the given SN has been received or not received, e.g. “0” is not received, ‘1’ is received. The non-received packets present in this list are included in the NACK. Note that non-received packets are removed from this list after expiry of the client buffering time (client_buffering_time) for the respective packet.  
      Step 2. The second step is to calculate the maximum number of retransmissions (max_no_rtxs) for the packets reported in the General NACK report blocks for the given RTCP bandwidth, client_rtcp_bandwidth. Since the packets pending for retransmission is the minimum set of information that every retransmission packet should comprise to enforce a minimum number of retransmissions (min_no_rtxs), the maximum number of retransmissions may be calculated by dividing the client_buffering_time′ through the current value of the RTCP report interval, rtcp_report_interval:  
                     max_no   ⁢   _rtxs     =       ⁢       client_buffering   ⁢     _time   ′         rtcp_report   ⁢   _interval                   =       ⁢     client_buffering   ⁢       _time   ′     ·                       ⁢       client_rtcp   ⁢           ⁢   _bandwidth       avg_rtcp   ⁢           ⁢   _paket   ⁢           ⁢     _size   NACK                       (   14   )             
 
      It should be noted that this calculation comprises the simplification provided in Equation (13) above. The RTCP report interval parameter, rtcp_report_interval, is commonly kept by each RTP client and server as part of the RTP basic algorithms.  
      The average RTCP packet size, avg_rtcp_packet_size NACK , may be calculated as (see also Equation 11 above): 
 
avg_rtcp_packet_size NACK = 1/16·current_rtcp_packet_size NACK + 1/16·avg_rtcp_packet_size  (15) 
 
      The current RTCP packet size (in bytes), i.e. the packet size of the next feedback message when comprising a General NACK report block (but no Loss RLE report block), may be calculated as: 
 
current_rtcp_packet_size NACK   =K +(step( n, 17))·4  (16) 
 
 where the step( ) is the mathematical step function that takes n and the constant 17 as parameters. 17 is the number of packets that can be requested in a single General NACK report block. It is of course possible to choose a different constant than 17 as a parameter for the step function in case more/less than 17 packets may be requested by a single General NACK report block. For several values of n: 
 
      step(n, 17)=
          0 if n&lt;=0;     1 if 1&lt;=n&lt;=17;     2 if 1·17+1&lt;=n&lt;=2·17;     3 if 2·17+1&lt;=n&lt;=3·17     . . .        

      K is constant accounting for the fixed fields in IP packet containing RTCP receiver messages with at least one General NACK message.  
      For example, if the packet has a standard composition of one Receiver Report (RR) of 32 bytes plus one SDES item (with CNAME) of 12 bytes without encryption header or profile specific extensions, the fixed fields amount to IP header+UDP header+RTCP header+1 RR+1 SDES item (CNAME)+NACK fixed fields, i.e. 20+8+8+24+12+12=84 bytes.  
      In case the packet contains profile specific extensions or other SDES items such as phone number, email or other RTCP report blocks such as a BYE packet or an application-specific packet (APP), then the fixed fields must be added to this constant K. Same applies for K′ below.  
      If the calculated maximum number of retransmissions (Equation 14), max_no_rtxs, is larger than one (&gt;1) this means that lost packets have a chance to be received in a later retransmission(s).  
      Typically the client may be required to comply with a minimum number of retransmission, min_no_rtxs. If this is not possible, i.e. max_no_rtxs&lt;min_no_rtxs, the client may decide to continue with a lower number of retransmissions or to re-initiate the session, choosing other settings (e.g. some codec configuration that requires less buffering time, by choosing a higher RTCP bandwidth, a stream that is better protected so that min_no_rtxs is smaller, etc.) as it is not possible to provide the required minimum number of retransmissions for packets within the constraints given by the RTCP bandwidth share and the client buffering time.  
      In other words, if there is a minimum number of retransmissions required by the client which can be met according to the calculations above, it is possible to add data to the feedback until the maximum packet size (see below) is reached. Otherwise the RTCP packets are too big and retransmissions cannot be requested as often as required.  
      The maximum size of the RTCP packet, max_rtcp_packet_size, that allows for requesting a number of min_no_rtxs retransmissions of a packet may be calculated as follows:  
                     max_rtcp   ⁢   _packet   ⁢   _size     =       ⁢     client_buffering   ⁢       _time   ′     ·                       ⁢       client_rtcp   ⁢           ⁢   _bandwidth       min_no   ⁢   _rtxs                     (   17   )             
 
      Note that the Equation 17 is defined for every value of min_no_rtxs, since the latter parameter is always greater than 1. Otherwise it would not make sense to use retransmissions as suggested by Rey et al. in the Internet Draft mentioned above.  
      Step 3. Assuming that the client is able to enforce min_no_rtxs and assuming that max_no_rtxs&gt;min_no_rtxs the next step is to fill the feedback messages with Loss RLE report blocks (in addition to General NACKs for requesting the retransmissions) until the max_rtcp_packet_size as calculated using Equation 17 is reached. In other words, the difference between the max_rtcp_packet_size and the current_rtcp_packet_size NACK  is equal to the amount of data that may be added to the current RTCP packet without violating the requirements for a minimum number of retransmissions, min_no_rtxs. 
 
payload_margin=max_rtcp_packet size−current_rtcp_packet_size NACK   (18) 
 
      Similar to the case above (see Equation 16), the general formula for calculation of the RTCP packet size, including IP and UDP headers, for the RTCP with General NACK messages and Loss RLE Reports as a function of n is as follows: 
 
current_rtcp_packet_size= K ′+(step( n ,17)+step′( n′ ,30))·4  (19) 
 
 where the step′( ) is the same as above and step( ) is the mathematical step function that takes n and the constant 30 as parameters. Please note that in Equation 19 it is assumed that at least Loss RLE report block is included in the feedback, otherwise Equation 16 may be used to calculate the current RTCP packet size. The parameter 30 is the number of packets that can be reported on in a single Loss RLE report block. It is of course possible to choose a different constant than 30 as a parameter for the step function in case more/less than 30 packets may be reported by a single RLE report block. For several values of no 
 
      step′(n′,30)=
          0 if n′&lt;=0;     1 if 1&lt;=n′&lt;=1·30;     2 if 1·30+1&lt;=n′&lt;=2·30;     3 if 2·30+1&lt;=n′&lt;= 3·30;       . . .        

      It should be noted that n and n′ increase step-wise. The reason for this is that the RTCP reports are 32-bit aligned and a whole 4-byte word is added whenever the number of packets to report exceeds a multiple of 17 or 30.)  
      In Equation 19 K′ is 6 bytes larger than K defined for Equation 16 above, since (at least) one Loss RLE report block included in the (current) RTCP feedback message to be sent next.  
      If the client is required to comply with a given report_redundancy value, it may include past Loss RLE report blocks as defined by this value. E.g. a redundancy_value of 2 means that the same Loss RLE report is sent in two consecutive RTCP reports. If the value is three, then the same Loss RLE report block is present in 3 consecutive RTCP packets and so on. In other words, the if the reporting redundancy is set to 2 a single RTCP report (i.e. the RLE report blocks therein) should report over the past two RTCP report intervals, if the reporting redundancy is set to 3 the past three RTCP report intervals should be reported on, etc. This may require that the client keeps a list (e.g. a two column array) of received and non-received packets, including packets that may be expired. This is different from the contents of the client buffer, which only includes packets that have not yet expired.  
      Step 4. It may also happen that including the required number of Loss RLE report blocks that would have to be added to the next RTCP feedback message (in addition to General NACK report block(s)) for providing the required reporting redundancy exceeds the allowed maximum RTCP packet size, max_rtcp_packet_size. In this case the client may try to reduce the size of the Loss RLE report block(s) by run-length encoding. As a further option other methods like thinning as described above may be used to reduce the packet size. However, when considering the use of thinning the reporting redundancy will be decreased.  
      If these mechanisms do not allow to sufficiently reduce the size of the RLE report blocks to guarantee the required reporting redundancy the client may decide to not comply with this report_redundancy value, and may continue sending. Alternatively, it may re-initiate the session and choose a larger client buffering time or some codec configuration with lower bitrate.  
      Step 5. Finally the rest of the variable update and standard procedures in the standard algorithms as per RFC 3550, Annex A, such as the avg_rtcp_packetsize and rtcp_report_interval, are calculated in the standard way:  
               avg_rtcp   ⁢           ⁢   _packet   ⁢           ⁢   _size     =       ⁢           1   /   16     ·   current_rtcp     ⁢   _packet   ⁢   _size     +                     ⁢         15   /   16     ·   avg_rtcp     ⁢   _packet   ⁢   _size               
         rtcp_report   ⁢   _interval     =       avg_rtcp   ⁢   _packet   ⁢   _size       client_rtcp   ⁢   _bandwidth           
 
      Every time an RTCP packet is sent the steps 1 through 5 may be repeated. Alternatively, the client may also choose to iterate through these steps only upon changes in the client capabilities, especially upon changes in the client buffering time or the client RTCP bandwidth occur.  
      After having sent the feedback message, the client may update the various parameters of the session according to the provisions outlined in RFC 3550. For example, the client may recalculate the average RTCP packet size, may recalculate the RTCP report interval resulting therefrom, may update the send buffer contents, and may start calculations for the next RTCP feedback to send.  
      In another embodiment, the size of the Loss RLE report block may be further reduced, by the client reducing the report_redundancy. However, in this situation the question arises whether the client may still obtain a reasonable level for redundant reporting.  
      Reducing the reporting redundancy may be possible in implementations that allow for measuring the packet loss rate on the link. The packet loss rate may for example be estimated from RTCP reports from the sender. Alternatively or in addition the client may have some a-priori knowledge about the link QoS from the link set-up procedure. Hence, when the packet loss rate on a particular link to the client is low/high, the level of redundant reporting may be further reduced/increased, since the lower/higher the packet loss rate the more likely/unlikely it is that the feedback of the client is received at the sender when using an unreliable transport protocol.  
      Another method to reduce the Loss RLE Report block size, apart from those specified in the Internet Draft of Friedman et al. may for example be based on the an estimation of the importance of individual packets. E.g. in progressive coding technologies such as MPEG-4 packets have different priorities. Less important packets, i.e. P-frames, may thus be dropped in the reporting.  
      Alternatively or in addition the current application level QoS and link level QoS may be taken into account for reducing the size of Loss RLE report blocks. E.g. if current application level QoS is sufficiently high and/or the link level QoS is low (i.e. congestion on the link), the client could decide to drop the reporting for some lost packets. For example, the MPEG4 codec implements advanced packet loss resilience measures. In some cases and depending on the content of the stream, it might well be the case that several packets were lost on the link, while the quality of the picture is not affected by these losses; in particular, if B frames are lost, the effect is minimum and the cost of a packet retransmission may be greater and so may the application decide to drop this packet.  
      The invention has the following advantages for 3GPP PSS unicast streaming services. First, it provides the 3GPP PSS framework with additional mechanism for streaming servers and clients for constructing the RTCP Receiver Reports using General NACKs and RLE Report blocks. Assuming the client knows the client buffering time and the client_rtcp_bandwidth, the method enables specifying the size of the reports to provide a minimum number of retransmissions, min_no_rtxs, and a given report_redundancy, how often should they be sent and how many packets they should cover.  
      As indicated above, report redundancy is important for charging applications and for network monitoring especially with respect to the performance of retransmissions, i.e. how many times packets are re-transmitted before they are (successfully) received by the client.  
      Further, the invention defines a framework for specifying a wide spectrum of client configurations, such that from a most simple client to a full featured client can be set-up depending on which information is available at the client and how efficient the Loss RLE Report compression should be. Moreover, some embodiments of the invention provide additional methods to reduce the size of the RTCP feedback.  
      Next, different and more specific exemplary embodiments of the invention will be described with reference to  FIGS. 4, 5  and  6 .  
       FIG. 4  shows the initial steps of a method for providing RTCP feedback within a streaming session according to an embodiment of the invention. Upon expiry of the current RTCP report interval the client prepares to send the next RTCP feedback to the server. First (steps  401  to  404 ) the client may determine whether the session parameters set allow to ensure that a required minimum number of retransmissions (min_no_rtxs) may be sent. For this purpose, the client may determine  401  the size of the current RTCP packet (next RTCP packet to sent) assuming it only contains the General NACK report block(s) necessary to ensure the minimum number of retransmissions configured for the session. As described earlier (see equation 16) the size of the current RTCP packet will depend on the number of packets to report on.  
      Based on this calculated value (current_rtcp_packet_size NACK ) as well as the average size of previous RTCP feedback messages (sent_rtcp_packet_size), the average RTCP packet size (avg_rtcp_packet_size NACK ) may be determined  402  using Equation 15 above. Next, the maximum number of retransmissions (max_no rtxs) that can be enabled 403 within the client buffering time is calculated (see Equation 14).  
      Based on these calculations, the client may determine in step  404 , whether the requested minimum number of minimum retransmissions (min_no_rtxs) can be enabled or not. If not, the client may reconfigure or restart ( 405 ) the session with a different set of session parameters as described above.  
      In case the max_no_trx&gt;mir_no_rtxs, the session parameters client buffering time, RTCP bandwidth share and the minimum number of retransmissions have been chosen such that the client may be operated within these constraint parameters. In a next step  406 , the client may therefore add the General NACK report block(s) necessary to provide a minimum number of retransmissions to the current feedback message. In case no (additional) reporting with Loss RLE report blocks is required for the session (see step  407 ) the client may proceed with transmitting  408  the current feedback message containing only General NACK report block(s) to the server.  
      If (redundant) reporting on packet loss is mandatory for the session, the client may next determine whether the current feedback message has enough capacity to comprise not only the General NACK report block(s) for ensuring the minimum number of retransmissions but additionally the number R of Loss RLE report block(s) required for ensuring the redundant reporting on packet loss.  
      For this purpose, the client may next determine the maximum size of the RTCP feedback messages that allow for providing a minimum number of retransmissions in step  409  (see Equation 17). As indicated above, the maximum RTCP packet size, max_rtcp_packet_size, is a measure indicating the average message size allowed for providing feedback when ensuring a minimum number of retransmissions.  
      Due to knowing that max_no_rbc&gt;min_no_rtxs (see step  404 ), it is also clear that the maximum RTCP packet size is equal to or larger than the average RTCP packet size containing the General NACK report block(s),  
      In step  410  the client determines the payload margin left in the current RTCP packet. In other words, the client determines how many bytes (payload_margin) may be further added to the current RTCP packet already comprising the General NACK report block(s) without violating the maximum allowable (average) size of the RTCP feedback.  
      Generally, the next steps following in the flow chart shown in  FIGS. 5 and 6  should be understood as examples illustrating how the remaining space (payload_margin) within a current RTCP feedback message may be filled with Loss RLE report block(s) to ensure a (minimum) redundant reporting, if possible at all. As will be explained in more detail below it may not be possible in some cases to enable a minimum number of redundant reports on lost packets. Nevertheless, in this situation, it may be desirable to provide at least some level of (redundant) reporting rather than no reporting at all.  
      In a next step  501  ( FIG. 5 ), the client may build a number R of Loss RLE report blocks which would be necessary for enabling the requested minimum reporting redundancy. As the size of the Loss RLE report blocks may be too large to fit them into the payload margin, the client may perform a run-length compression of the Loss RLE report blocks as illustrated by step  502 . Please note that this step is optional. Next, the size of the R (compressed) Loss RLE report block is determined in step  503  and the value is compared with the payload margin in step  504  in order to determine whether all R Loss RLE report blocks fit into the current RTCP feedback message without violating the maximum allowable (average) RTCP packet size that allows to enable a requested minimum number of retransmissions.  
      If so, the R (compressed) Loss RLE report blocks are added  505  to the feedback message and the client transmits  506  a RTCP feedback message comprising a number of General NACK report blocks and a number (R) of Loss RLE report blocks. In this case, the client may ensure that the minimum number of retransmissions is enabled and also a minimum reporting redundancy can be provided.  
      If however the size of the R (compressed) Loss RLE report blocks in step  504  is larger than the payload margin, the client may try to provide at least some (redundant) reporting on lost packets. For this purpose, the client may further proceed with step  507  and may reduce the number of Loss RLE report blocks to be included in the current feedback until the remaining number of (compressed) Loss RLE report blocks fits  509  into the payload margin of the current RTCP feedback message. For example, the client may consecutively drop (see steps  507 ,  508 ,  509 ,  511 ) report blocks reporting on the “oldest” packets, as same may have been already been reported in previous reports. Another possibility may be to perform thinning until the size of the Loss RLE report blocks drops below the payload margin. Event further, the client may also take into account the importance of packets reported on as has been described above. Another alternative may be to combine these mechanisms as needed.  
      In this respect  FIG. 6  illustrates further steps of forming a RTCP feedback message according to another embodiment of the invention. The steps similar to those in  FIG. 5  have been assigned the same reference numerals and their description is omitted for brevity in the following.  
      The steps illustrated in  FIG. 6  allow combining thinning and other mechanisms to reduce the size of the Loss RLE report blocks. Important to notice is that the decision on whether thinning (see step  603 ) is applied may be based on the Quality of Service (QoS) that is presently available (see steps  601  and  602 ) on downlink to the client.  
      E.g. the calculated values of the “fraction lost” field within a Sender and/or Receiver report block of the RTCP feedback messages (see RFC 3500, section 6.4, may be evaluated and taken as an indication of the present QoS on the uplink and downlink, respectively. Alternatively, also explicitly negotiated QoS measurements may be provided from the mobile communication system. If for example the indicated QoS on the uplink is high, i.e. this may mean that it is likely that the RTCP feedback of the client is not lost though transmitted using an unreliable transport protocol. Thus, the client may decide that a lower reporting redundancy than configured by the server may be sufficient. In this case, the client may decide to apply thinning to the Loss RLE report blocks. Thus, step  605  is similar to step  505  in  FIG. 5  except for the client may no longer guarantee a minimum reporting redundancy if thinning has been applied.  
      In another embodiment of the invention, the client may even decide to reduce the minimum number of retransmissions based on the QoS available. In this case, the client could consider not to include all General NACK report blocks to the feedback but to only report on the latest, newer packets in a manner similar as described for steps  507 ,  508 ,  509  and  511  of  FIG. 5 .  
      Another embodiment of the invention relates to the implementation of the above described various embodiments of the invention. It is recognized that the various above mentioned methods as well as the various logical blocks of the figures described above may be implemented or performed using computing devices (processors), as for example general purpose processors, digital signal processors (DSP), application specific integrated circuits (ASIC), field programmable gate arrays (FPGA) or other programmable logic devices, etc. The various embodiments of the invention may also be performed or embodied by a combination of these devices.  
      Further, the various embodiments of the invention, variations thereof and solutions for QoS-aware scheduling may also be implemented by means of software modules which are executed by a processor or directly in hardware. Also a combination of software modules and a hardware implementation may be possible. The software modules may be stored on any kind of computer readable storage media, for example RAM, EPROM, EEPROM, flash memory, registers, hard disks, CD-ROM, DVD, etc.