WO2002091691A1

WO2002091691A1 - Multiplexed coding

Info

Publication number: WO2002091691A1
Application number: PCT/EP2002/004713
Authority: WO
Inventors: Sören Vang ANDERSEN; Alan Duric; Gernot Kubin
Original assignee: Global Ip Sound Ab
Priority date: 2001-05-03
Filing date: 2002-04-29
Publication date: 2002-11-14
Also published as: SE0101550D0

Abstract

The present invention relates to a signal processing in connection with digital signal transmission over, and reception from, a packet switched network. The invention is applicable when a set of digital signals share a common transmission path. The information about a segment of each digital signal among the set of digital signals is distributed in several data packets. Data packet payloads are assembled such way that each payload includes a part of the segment representation of several segments of a set of multiple signal segments, i.e. any data packet will contain a part of the complete transmitted information about several ones of the multiple signal segments. Effects on a reconstructed signal due to lost packets are alleviated since a lost packet only leads to a partial loss of information for each segment of a set of multiple segments.

Description

MULTIPLEXED CODING

Technical Field of the Invention

The present invention relates to signal processing in connection with digital signal transmission over, and reception from, a packet switched network.

Technical Background and Prior Art

Real-time transmissions over packet switched networks, such as speech, audio or video over Internet Protocol based networks (mainly the Internet or Intranet networks) , has become increasingly attractive due to a number of features . These features include such things as relatively low operating costs, easy integration of new services, and one network for both non-real-time and real-time data. Real-time data, typically a speech, an audio or a video signal, are in packet switched systems converted into a digital signal, i.e. into a bitstream, which is divided in portions of suitable size in order to be transmitted in data packets over the packet switched network from a transmitter end to a receiver end. As packet switched networks originally were designed for transmission of non-real-time data, transmissions of real-time data over such networks causes some problems. Data packets can be lost during transmission, as they can be deliberately discarded by the network due to congestion problems or transmission errors. In non-realtime applications this is not a problem since a lost packet can be retransmitted. However, retransmission is not a possible solution for real-time applications that are delay sensitive. A packet that arrives too late to a real-time application cannot be used to reconstruct the corresponding signal since this signal already has been, or should have been, delivered to the receiving end, e.g. for playback by a speaker or for visualisation on a display screen. Therefore, a packet that arrives too late is equivalent to a lost packet.

When transferring a real-time signal as packets, the main problem with lost or delayed data packets is the introduction of distortion in the reconstructed signal. The distortion results from the fact that signal segments conveyed by lost or delayed data packets cannot be reconstructed .

When transferring a signal it is most often desired to use as little bandwidth as possible. As is well known, many signals have patterns containing redundancies. Appropriate coding methods can avoid the transmission of the redundant information thereby enabling a more bandwidth effective transmission of the signal . Typical coding methods taking advantage of such redundancies are predictive coding methods. A predictive coding method encodes a signal pattern based on dependencies between the pattern representations. It encodes the signal for transmission with a fixed bit rate and with a trade-off between the signal quality and the transmitted bit rate. Examples of predictive coding methods used for speech are Adaptive Predictive Coding (APC) and Code Excited Linear Prediction (CELP) , which both coding methods are well known to a person skilled in the art. To alleviate the effects that lost packets have on the reconstructed signal with respect to lost signal information, two classes of signal processing methods have been proposed in the prior art . A first class adds redundancy and significant delay at the transmitter side, e.g., with loss-resilient codes or with multiple- description source/channel coding. A second class attempts perceptual concealment of the packet loss through signal interpolation at the receiver side. Thus, the drawbacks with these methods are either increased network payload data rate and increased transmission delay, or degraded perceptual quality of the received signal, or both.

ω t to μ> o H LΠ LΠ σ LΠ o Π

interconnects two circuit-switched networks with gateways performing dedicated point-to-point communications.

As another example, the path sharing traffic can be the path between a base station and another node, such as a Mobile Switching Centre, in a mobile communication system, in which the sources or destinations are Mobile Stations, or other radio transceivers, connected to the base station via radio links.

Yet other applications are so called streaming and multicasting in packet data networks.

As mentioned, the invention is also applicable when traffic share the same connection between two interconnected routers, or communication nodes, in which case the paths are separated before and after those routers.

Furthermore, traffic may be transmitted from the same gateway or telecommunications switch and share the same path to a router. Correspondingly, traffic may be received by the same gateway or telecommunications switch after having shared the same path from a router.

The multiplexing is preferably performed by multiplexing representations, e.g. quantized representations, from each one of the multiple segments, i.e. from each segment of a set of segments, or by multiplexing parts from each one of the multiple segments based on linear combinations of digital samples from all of the multiple segments.

According to an embodiment, loss-resilient codes are added to the segments in order to be able to regenerate parts of segments that have been lost due to packet losses. These loss-resilient codes further alleviate the . effect of packet losses in a system operating in accordance with the present invention.

Thus, the present invention is suitable for use in packet switched networks were packet loss may occur. The invention provides robust coding with no redundancy overhead, even though redundancy can be added in a L > t H μ>

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should by no means be interpreted as limiting on the scope of the claims .

The above mentioned and further features of, and advantages with, the present invention, will be more fully understood from the following description, with reference to the accompanying drawings, of exemplifying embodiments thereof .

Brief Description of the Drawings Exemplifying embodiments of the present invention will be described by way of example with reference to the accompanying drawings, in which:

Fig. 1 shows the basic principle of the multiplexing scheme in accordance with an embodiment of the invention in an exemplified, and schematically illustrated, system; Fig. 2 shows two Media Gateways adapted for transmitting and receiving multiplexed segment representations in accordance with an embodiment of the invention; and Fig. 3 shows a multiplexed encoding arrangement transmitting multiple segments to a multiplexed decoding arrangements, in accordance with embodiments of the invention.

Detailed Description of the Invention

With reference to Fig. 1 the principle of multiplexing in accordance with an embodiment of the invention is shown. A first communication node (Media Gateway) 100 receives a set of multiple segments. Each segment belongs to a respective encoded real-time signal. For ease of description, only three signal S' , S" and S'" with respective segments are depicted in Fig. 1, however, any number (greater than one) of signals can be subject to the practising of the present invention. Furthermore, for ease of description, each segment includes a very limited number of elements in its representation. All the three signals share a common path from the first OJ J t IS μ> μ¹

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In Fig. 1 the incoming segments of signals S' , S" and S'" to the Media Gateway 100 are either already encoded by their respective sources, or they are encoded by the Media Gateway 100. In Fig. 1 it can also be seen that if a data packet and its payload is lost, every third part of a segment representation will be lost. To further alleviate the effect of lost data packets, loss resilient coding may advantageously be added to the multiplexing scheme. The resulting redundancy bits ( r , r₂ and r₃ ) of such loss resilient coding have been included in the segments and packet payloads shown in Fig. 1. Again, the number of redundancy bits indicated in each segment/payload is merely chosen for ease of description, any number of redundancy bits can be used. Moreover, the loss resilient coding adding the redundancy bits is in Fig. 1 indicated as having been performed by the respective sources, however, this can alternatively be performed by the Media Gateway 100. The loss resilient coding is preferably added by applying a coding method appreciated by a person skilled in the art, such as Reed-Solomon coding. Furthermore, a person skilled in the art will also appreciate that by means of the segment parts and redundancy bits that are received by Media Gateway 110, lost segment representation parts in a lost packet payload can be reconstructed.

With reference to Fig. 2, two Media Gateways, or communication nodes, at respective ends of a path shared by multiple connections are shown. Each Media Gateway (MG) is controlled by its respective Media Gateway

Controller (MGC) over a Media Gateway Control Protocol (MGCP) . The MGCP is in essence a master/slave protocol where the Media Gateway executes the commands that are sent by the Media Gateway Controller. The transmitting Media Gateway 200 is controlled by Media Gateway Controller 210 and includes a MUX server function 220. The MUX server function is vendor specific and does not effect other architecture elements or any standardized protocol. Correspondingly, the receiving Media Gateway 250 is controlled by Media Gateway Controller 260 and includes a vendor specific DeMUX server function 270.

Inputted to each Coder 230 at the transmitting MG 200 are multiple real-time data streams. Upon setting up a multiplexed connection between the two MG' s 200 and 250, the MGC 210 of MG 200 will issue a call set-up to MGC 260 of MG 250. The MUX server function 220 includes a table of what coders that are included by MG 200 and the destination of their encoded data streams. This table is built by listening to the signalling between the MGC 210 and MG 200 when setting up the Coders 230 and the destinations of their respective data streams.

The MUX server function 220 is implemented as a process which receives segment parts outputted from the Coder 230 processes. Using its included table the MUX server function 220 multiplexes the segment parts received from the different Coders 230. When a data packet payload has been assembled, this payload will be outputted to one of the waiting Coder processes. This Coder will in turn output the assembled payload to a Packetizer 240. In case the path between the two MG's is over an Internet Protocol network, the Packetizer 240 will add IP headers etc. to the packet streams.

Correspondingly, upon reception of the call set-up from MGC 210, signalling will occur between the MGC 260 and the Media Gateway 250. The DeMUX server function 270 will listen to this signalling and build a table to be used for de-multiplexing of received data packet payloads that includes multiplexed segment representation parts. When packet payloads are received by the Decoders 280 from a Depacketizer 290, these payloads will be outputted to the DeMUX server function process 270. After the DeMUX process has disassembled a number of payloads and extracted segment representation parts therefrom, parts that together form a complete segment will be transferred as a full segment representation to one of the waiting Decoders 280 for further transmission to its destination.

A person skilled in the art will appreciate other ways of implementing the assembling/disassembling of packet payloads, e.g. by having the MUX server function 220 and DeMUX server function 270 interact with the Packetizer 240 and Depacketizer 290, respectively.

Thus, in Fig. 2, Media Gateway 200 and Media Gateway 250 act as a first and a second communication node, respectively. Media Gateway 200 encodes and assembles multiple segments and Media Gateway 250 performs disassembling and decoding with respect to the segments. Alternatively, the encoding performed by the set of Coders 230 could be located at the respective signal ^" sources and the decoding performed by the set of Decoders 280 at the respective signal destinations.

With reference to Fig. 3, the multiplexing/demultiplexing of signal segment parts in accordance with exemplifying embodiments of the invention will be described in greater detail .

In Fig. 3 the system is based on K scalar adaptive predictive speech coders with noise feedback coding of the kind proposed by Atal and Schroeder in "Predictive coding of speech signals and subjective error criteria", B.S. Atal and M.R. Schroeder, IEEE Trans. Acoust . Speech Signal Processing, vol. 27, no. 3, pp. 247-254, 1979. In the figure only the k-. th predictive encoder and decoder are shown. In this system the traditional single-input single-output scalar quantizer have been replaced with what is herein referred to as a multiplexed quantizer 300. The multiplexed quantizer 300 takes at each sampling instant n the K inputs

to q from all K predictive encoders and outputs quantized representations

to qf_t back to the individual predictive encoders. In doing this, the multiplexed quantizer 300 generates K quantization indices to t,f each in the range 1 to 2^b where b is the number of bits allocated for each index. These indices are packetized and transmitted over the packet-switched network in K independent packet streams 310..320. At the decoder side, the received indices t,,¹ to i_n ^κ are available. These indices differ from the indices to i in the encoder whenever an index has been lost with a data packet on the network, in which case the index value is replaced by zero. The demultiplexer 330 resolves K representations

of the quantized information from the available indices z_n ¹ to i_n ^κ . These representations are subsequently input to the K predictive decoders. The resulting encoding and decoding system is shown in Figure 1. Not included in this figure is the LPC analysis and side information quantization which leads to coefficients for the predictors P (z) , noise-feedback filters F (z) , and scaling factors σ* for k = 1 . . K. The side information is conveniently encoded with a loss-resilient coding such as the Reed-Solomon code to enable transmission of this information in a robust manner using the same K packet streams as the multiplexed quantized information. The Reed-Solomon code is described in "Reed-Solomon Codes and their Applications", S.B. Wicker and V.K. Bhargava, IEEE Press, New York, 1994. With further reference to Fig. 3, three different multiplexing schemes in accordance with different embodiments of the invention will now be described: packet hopping, Hadamard multiplexing, and an extension of the Hadamard multiplexing that exploits a nonlinear preprocessing and estimation method.

In a packet hopping scheme the multiplexed quantizer block is a system in which indices from scalar quantizers are hopped, i.e., cycled from one packet stream to the next as the sample instant n increments . One version of this system is specified by the equations

= Q ( q^^mod^^+l ) (1)

and

~(n_+k)mod(K)₊l _{= Q}-ι₍ξ*_{} (3)}

for k = 1.. K. Here Q ( ^• ) and Q^"1 ( • ) are the mappings from quantizer input to quantization index and from quantization index to the quantized representation of the quantizer input, respectively. For an adequate response to packet losses Q^_:1"(0) = 0. The notation (•)mod(iⁱ) denotes the modulo K operation.

The packet hopping described above can be expressed as a particular orthogonal transformation of the input to the multiplexed quantizer followed by a quantization of the transform output. If we define column vectors with elements equal to the K scalar input, output, or index values for the multiplexed quantizer and demultiplexer, such that e.g.,

Then the multiplexed quantizer is defined by the following equations :

c„ = M„q„

and The demultiplexer on the receiver side is defined by:

and

q„ = M^c„

The equivalence of these equations with the packet hopping described by Equations 1 to 3 is obtained by letting the transform matrix M„ equate an adequate time varying row or column permutation of an identity matrix. With this formulation of the multiplexing, it is relevant to introduce, in alternative embodiments, other transform matrices than the row or column permuted identity matrix. A simple, yet relevant, transform for this purpose is the normalized Hadamard transform disclosed in "Orthogonal Designs; Quadratic Forms and

Hadamard Matrices", A.V. Geramita and J. Seberry, Marcel Dekker, 1979. The Hadamard multiplexing will hold advantages over the packet hopping method. These advantages are explained in the following. One advantage is that whenever less than K packets are lost in the network there are no full erasures of any sample in any of the quantized prediction error signals. This advantage can be exploited when the Hadamard transform is combined with a nonlinear preprocessing and estimation scheme as described below.

Let us assume the elements of q_n to be uncorrelated and neglect the impact of quantization noise. Then the matrix M^ is the linear minimum mean-squared error estimator for q_M given the coefficient vector c„ . This estimator is the mean-square optimum for Gaussian q_n . However, the gain-scaled linear prediction errors for voiced speech signals are known to be non-Gaussian. Thus, a nonlinear estimator can result in lower mean-squared error. Indeed, we observed in preliminary experiments that nonlinear estimation could lead to a significant decrease of the mean-squared error. However, the nonlinear estimation led to very high computational complexity. Therefore we adopt an alternative method in which a well defined nonlinearity is applied to the input of the multiplexed quantizer. Knowledge of this nonlinearity can then subsequently be exploited to improve the reconstructed quantized prediction errors in the case of packet losses.

The general method suitable to use is to zero K_z of the K inputs to the multiplexed quantizer prior to applying the transform M_B . Advantageously, the K_z inputs with lowest amplitudes are set to zero. In the method, no information about the position of zero valued elements in q_n is conveyed to the decoder, only the knowledge that K_Z of the elements were zero is exploited. The method is described below. Suppose that the number of lost packet streams is K_\ps and the lost packet streams are indexed by an integer set k, . Then we may formulate a set of equations relating to the received coefficients c_n with the encoded scaled prediction errors q_n .

q_n = M^T _{n n} + M ( ,k,Aa (4)

In this equation a is a K\_ps dimensional unknown vector. We have used a matlab-style notation M^(:,k_;jM) to denote a matrix consisting of the columns of M^ that are indexed

Now assume that q_n had X_lps zero-valued elements indexed by the set k_zi : q_n ( k_z. ) = 0 , then

provided that M^ ( k_Zi. , k_lps ) has full rank . Furthermore ,

« = c„ ( k^ ) ( 5 )

The indexing for the zero-valued elements of q_H is not known by the decoder, however there are

ways in which the decoder can assume K_\ps elements of q_n to be zero. Of these

will be true assumptions . Whenever i_zi > Kχ_ps there are multiple true assumptions and all true assumptions will result in the same value for a , i.e., the one given in Equation 5. Thus, the method applicable in the decoder is to calculate a for all C_x possible choices of k_z;. and select the a vector that occurred C₂ times. Hereafter q_n is obtained as the right-hand side of Equation 4.

Rank deficiency of the matrix M^(k_z.,k_//ω) limits the use of this method. For example, when M_ft equals the permuted identity matrix that follows from the packet hopping our nonlinear preprocessing and estimation does not apply. In contrast, when M„ is the normalized Hadamard transform, the method applies with no complications for i_ps = 1. For K\_vs > 1, rank deficiency can occur for some of the possible choices of k_z/ . In this case heuristics must be introduced in the selection of a .

To verify the benefits of the present invention, a coding experiment was coducted in which 12 speech files, each containing two utterances, were jointly encoded by multiplexed predictive coders (K = 12) . The speech files were encoded at 40 kbps and decoded after that a percentage of the data packets had been randomly dropped. Random packet loss rates between 0% and 40% were simulated. As a reference system, a 64 kbps /-law quantization and packet loss concealment (PLC) according to the ITU-G.711 standard was used. The reference system was simulated for the same speech files and packet losses as the multiplexed predictive coders.

A preference test was conducted of the 40 kbps multiplexed predictive coder with packet hopping versus the reference system. In this test listeners were subjected to 24 utterances processed by the two systems in randomized order. For each utterance the listeners made a preference decision. The results are given in Table 3.

Packet loss rate 0% 10^! 20% 30^! 40%

Preference to MPC 35% 63% 77% 92% 97% Preference to G.711 65% 37% 23% 8% 3%

Table 3.

We see from these results that multiplexed predictive coding was preferred over G.711 PLC for packet loss rates in the range from 10% to 40%. Thus, this implies that the multiplexed predictive coding system is more robust to packet losses than scalar quantization and packet loss concealment according to the G.711 standard. In addition, the multiplexed predictive coders can be designed to operate at significantly lower bit rates than the G.711 standard.

Claims

1. A method of signal processing in connection with digital signal transmission over a packet switched network, including the steps of: encoding a set of segments of respective digital signals into a set of corresponding segment representations; and assembling a set of data packet payloads for transmission in respective data packets over the packet switched network, wherein each data packet payload, for several segments of said set of segments, includes a part of the full segment representation, in such way that all parts of a segment that are included by respective data packet payloads together form the full representation of said a segment .

2. The method as claimed in claim 1, wherein the number of data packet payloads in said set of data packet payloads either is equal to, or different from, the number of segments in said set of segments.

3. The method as claimed in claims 1 or 2, said encoding step including encoding all segments of said set of segments essentially simultaneously.

4. The method as claimed in any one of claims 1 - 3, wherein said encoding step includes, with respect to each segment, quantization of digital signal samples to quantized representations of the samples, wherein the representation of the segment consists of quantized representations of the digital samples of the segment .

5. The method as claimed in any one of claims 1 - 4, wherein said part of the full segment representation is included in the data packet payload as quantized representations of corresponding digital signal samples.

6. The method as claimed in any one of claims 1 - 3, wherein the full segment representations of the segments of said set of segments include respective parts which are included in said set of data packet payloads as linear combinations of respective sets of digital samples, wherein each set of digital samples includes digital samples from all segments of said set of segments .

7. The method as claimed in any one of claims 1 - 3 or 6, said step of encoding multiple segments including the steps of: transforming a set of digital samples including digital samples from all segments of said set of segments into a set of linear combinations of these digital samples; quantizing the set of linear combinations to a corresponding set of quantized linear combinations, wherein the quantized linear combinations are provided to be included in respective data packet payloads as parts of full segment representations of the segments of said set of segments.

8. The method as claimed in claim 7, wherein the digital signal samples of said transforming step are transformed by means of a Hadamard transform.

9. The method as claimed in claim 7 or 8 , wherein said transforming step is preceded by the step of applying a predefined non-linearity with respect to the digital samples.

10. The method as claimed in claim 9, wherein said step of applying a non-linearity includes setting a number of those digital samples that have the lowest amplitude, among said set of digital samples, to the value zero before performing said transforming step.

11. The method as claimed in any one of the previous claims, wherein the encoding step includes the step of: applying a loss-resilient encoding of said set of segments; and adding redundancy bits obtained from the loss- resilient encoding to the segments of said set of segments, wherein also the redundancy bits are subject to the assembling step.

12. The method as claimed in any one of claims 1 - 11, wherein a first communication node performs said encoding step and said assembling step with respect to digital signals that are to be transmitted to the same second communication node, which digital signals are received from respective sources being directly or indirectly connected to the first communication node.

13. The method as claimed in claim 12, wherein said first communication node is a gateway or a telecommunications switch and said second communication node a gateway, a telecommunications switch, or a router in a telecommunications network.

14. The method as claimed in any one of claims 1 - 11, wherein said encoding step collectively is performed by a set of sources, each source encoding the segment of a corresponding digital signal for further transmission to a first communication node; and said assembling step is performed by said first communication node with respect to digital signals that are received from said set of sources and that are to be transmitted to the same second communication node.

15. The method as claimed in claim 14, wherein said first communication node is a gateway, a telecommunications switch, or a router in a telecommunications network and said second communication node a gateway, a telecommunications switch, or a router in a telecommunications network.

16. The method as claimed in claim 12 or 14, wherein said sources are mobile communication stations, said first communication node a base station in a radio communications network, and said second communication node a node in a radio communications network with which the base station communicates.

17. The method as claimed in any one of claims 1 - 16, wherein a digital signal corresponds to any real-time data traffic signal, such as a speech, audio or video signal .

18. An apparatus for processing digital signals to be transmitted over a packet switched network, wherein the apparatus includes means for performing the steps of the method as claimed in any one of claims 1-17.

19. A computer-readable medium storing computer- executable components for processing digital signals to be transmitted over a packet switched network, wherein the computer-executable components perform the steps of the method as claimed in any one of claims 1-17.

20. A method of signal processing in connection with digital signal reception from a packet switched network, including the steps of : disassembling a set of data packet payloads received in respective data packets from the packet switched network into a set of segment representations corresponding to a set of segments of respective digital signals, wherein, for several segments of said set of segments, a part of the full representation of the segment is extracted from each data packet payload, whereby all parts extracted with respect to a segment forms the full representation of said a segment; and decoding said set of segment representations into the corresponding set of segments .

21. The method as claimed in claim 20, wherein the number of data packet payloads in said set of data packet payloads either is equal to, or different from, the number of segments in said set of segments.

22. The method as claimed in claim 20 or 21, said decoding step including decoding all segment representations of said set of segment representations essentially simultaneously.

23. The method as claimed in any one of claims 20 -

22, wherein said decoding step includes, with respect to each segment, which consists of quantized representations of digital signal samples, dequantization of the sample representations to corresponding digital signal samples .

24. The method as claimed in any one of claims 20 -

23, wherein said part of the full representation of the segment included in each data packet payload includes quantized representations of corresponding digital signal samples .

25. The method as claimed in any one of claims 20 - 22, wherein parts of respective segment representations are extracted from said set of data packet payloads as linear combinations of a set of digital samples which includes samples from all segments of said set of segments .

26. The method as claimed in any one of claims 20 21 or 25, said step of decoding including the steps of: dequantizing parts of respective segment representations extracted from said set of data packet payloads to a set of linear combinations; and transforming said set of linear combinations to a set of digital samples which includes samples from all segments of said set of segments.

27. The method as claimed in claim 26, wherein the digital signal samples of said transforming step are transformed by means of a transpose matrix of a Hadamard transform.

28. The method as claimed in claim 26 or 27, wherein said transforming step is followed by the step of reconstructing lost linear combinations of said set of linear combinations, the reconstruction being based on knowledge that a number of the digital samples from which the set of linear combinations was derived at encoding was set to zero during encoding.

29. The method as claimed in any one of the previous claims, wherein the decoding step includes the step of exploiting redundancy bits included by the segment representations for recovering lost parts of the segment representations.

30. The method as claimed in any one of claims 20 - 29, wherein a second communication node performs said disassembling step and said decoding step with respect to packetized digital signals that are received from the same first communication node, which digital signals then are transferred to respective terminating entities being directly or indirectly connected to the second communication node.

31. The method as claimed in claim 30, wherein said second communication node is a gateway or a telecommunications switch and said first communication node a gateway, a telecommunications switch, or a router in a telecommunications network.

32. The method as claimed in any one of claims 20 - 29, wherein said disassembling step is performed by a second communication node with respect packetized digital signals that are received from the same first communication node, which digital signals then are transmitted to respective entities; and said decoding step collectively is performed by said entities, each entity decoding the segment representation of a corresponding digital signal .

33. The method as claimed in claim 32, wherein said second communication node is a gateway, a telecommunications switch, or a router in a telecommunications network and said first communication node a gateway, a telecommunications switch, or a router in a telecommunications network.

34. The method as claimed in claim 30 or 32, wherein said entities are mobile communication stations, said second communication node a base station in a radio communications network, and said first communication node a node in a radio communications network with which the base station communicates.

35. The method as claimed in any one of claims 20 - 34, wherein a digital signal corresponds to any real-time data traffic signal, such as a speech, audio or video signal .

36. An apparatus for processing digital signals received from a packet switched network, wherein the apparatus includes means for performing the steps of the method as claimed in any one of claims 20-35.

37. A computer-readable medium storing computer- executable components for processing digital signals received from a packet switched network, wherein the computer-executable components perform the steps of the method as claimed in any one of claims 20-35.