CN1618233A - Video conferencing system and method of operation - Google Patents

Abstract

A method of relaying video images in a multimedia videoconference between a plurality of multimedia user equipment (550, 560, 570, 580) includes the step of transmitting layered video images by a number of said plurality of user equipment wherein said layered video images inlcude a base layer (552, 562, 572, 582) and one or more enhancement layers (555, 565, 575, 585). The transmitted layered video images are received at a multipoint control unit (520), where a number of base layer video images of a number of active speakers (535) and one or more enhancement layers (540) of a most active speaker are selected. the multipoint control unit (520) transmits the base layer video images, and one or more enhancement layers (540) of the most active speaker, to one or more of the plurality of multimedia user equipment (550, 560, 570, 580). The identification of the speakers is much improved compared to traditional videoconference systems, as the available bandwidth is shared to allow one enhancement layer and several base layers to be sent, instead of only one full quality video stream.

Description

Video conferencing system and method of operation

technical field

The present invention relates to video conferencing. The present invention can be applied to a video exchange mechanism using layered video coding in centralized video conferencing based on H.323 and/or SIP, but is not limited thereto.

Background technique

As the pace of commerce increases and business relationships expand worldwide, the need to quickly and economically span communication distances becomes a major challenge. In order to succeed in an increasingly competitive marketplace, effectively bringing customers and staff together is key. Merchants are looking for flexible solutions to support real-time information sharing across borders and continents using various communication methods such as voice, video, image data and any combination thereof.

In particular, multinational organizations are increasingly looking to eliminate costly travel and connect multiple locations so groups within the organization can communicate more effectively. Multipoint conferencing systems operating on Internet Protocol (IP) networks attempt to address this need. In the field of the invention, it is known that in a multipoint video conference terminals can exchange audio and video streams in real time. The existing method of establishing a multipoint conference on an IP network is to use a multipoint control unit (MCU). An MCU is an endpoint on a network that provides the ability for three or more endpoints and/or communication gateways to participate in a multipoint conference. The MCU can also connect two terminals in a point-to-point conference so that they can develop into a multi-point conference.

Referring first to FIG. 1 , a known centralized conferencing model 100 is shown. Centralized conferences utilize MCU-based conference bridges. All terminals (endpoints) 120 , 122 , 125 send and receive media information in the form of audio, video and/or data signals and control information stream 140 to/from MCU 110 . These transfers can be done on a peer-to-peer basis. This is shown in Figure 1.

MCU100 consists of a multipoint controller (MC) and zero or more multipoint processors (MP). The MC handles call setup and call signaling negotiations between all terminals to determine common capabilities for audio and video processing. MC110 does not directly handle any one media stream. This is left to the MP to mix, switch and process audio, video and/or data bits.

In this way, the MCU provides the capability to hold multi-location conferences, sales meetings, group meetings and other 'face-to-face' communications. Multipoint conferencing is known to be used in various applications such as:

(i) Executives and managers in multiple locations are able to meet 'face-to-face', share real-time information, and make decisions faster without any loss of time, expense, and need for travel.

(ii) project teams and knowledge workers can coordinate their respective tasks in real-time and view and revise shared documents, manuscripts, designs and files; and

(iii) Students, trainees and employees at remote locations can access shared educational/training resources across any distance or time zone.

Therefore, it is conceivable that MCU-based systems will play an important role in future multimedia communications over IP-based networks.

Such multimedia communications typically employ video transmission. In this type of transmission, image sequences, usually referred to as frames, are transferred between the sending and receiving units. Various methods can be used to establish a multi-point multimedia conference system, such as the methods specified in the H.232 and SIP session layer protocol standards. References on SIP can be found at:

http://www.ietf.org/rfc/rfc2534.txt , and

http://www.cs.columbia.edu/~hgs/sip .

Furthermore, for example in systems using ITU H.263 video compression [ITU-T Recommendation, H.263, 'Video Coding for Low Bit Rate Communication'], the first frame of a video sequence includes a considerable amount of synthetic image data, usually It is called intra-frame coding information. Since the intra-coded frame is the first frame, it can provide a substantial portion of the image to be displayed. Intra-coded frames are followed by inter-coded (predicted) information, which typically includes data related to changes in the picture being sent. Thus predicted inter-coded information contains much less information than intra-coded information.

In a traditional multimedia conferencing system, users need to identify themselves when they speak, so that the receiving terminal knows who is speaking. Obviously, if the sending terminal does not recognize itself, the listening user will have to guess who is speaking.

One known technique solves this problem by analyzing the audio stream and forwarding the active speaker's name and video stream to all parties involved. In a centralized conferencing system, the MCU usually performs this function. The MCU can then send the speaker's name and corresponding video and audio streams to all participants by switching the appropriate input multimedia streams to the input ports/paths.

Video switching is a well-known technique that aims to deliver to each endpoint a separate video stream, equivalent to arranging multiple point-to-point sessions. Video exchanges can be:

(i) Voice-activated switching, where the MCU sends video of the active speaker.

(ii) A time-activated exchange, where each participant's video is sent sequentially at predetermined time intervals.

(iii) Personal video opt-in exchange, where each endpoint can request which party video streams he/she wishes to receive.

Referring now to FIG. 2, a functional block diagram of a conventional video switching mechanism 200 is shown. In a traditional centralized conference system, video exchange is performed as follows. MCU 220 , such as MCU 200 located within Internet Protocol (IP) based network 210 , includes switch 230 . The MCU 220 receives the video streams 255 , 265 , 275 , 285 of all participants (user devices) 250 , 260 , 270 , 280 . The MCU may also separately receive the combined (multiplexed) audio stream 290 from the speaking party. Afterwards, the MCU 220 selects a video stream and sends the video stream 240 to all participants 250 , 260 , 270 , 280 .

One downside of this traditional system is that they only send the video stream of the active speaker. Users may also have an issue identifying the speaker of the video stream if several speakers are speaking at the same time or the active speaker keeps changing. This is especially problematic in large video conferences.

Alternatively, each participant's video may be sent to all participants. However, in wireless based conferencing this method suffers due to bandwidth limitations.

In the field of video technology, it is known that video is sent as a series of still images/pictures. Because the quality of a video signal is affected during video signal encoding and compression, it is known to include additional 'layers' of information based on the differences between the video signal and the encoded video bitstream. The inclusion of additional layers enables the received signal quality to be enhanced with decoding and/or decompression. Therefore, a hierarchical video bitstream is generated using a hierarchical structure of pictures and enhanced pictures divided into one or more layers.

In a layered (scalable) video bitstream, the video signal can be enhanced outside the base layer by one of the following:

(i) Improve the resolution of the picture (spatial scalability);

(ii) include error messages to improve the signal-to-noise ratio (SNR measurability) of the picture; or

(iii) Include extra frames to increase frame rate (temporal scalability).

Such enhancements can be applied to the entire frame, or to objects of arbitrary shape in the frame, which is called object-based scalability. In order to preserve the discretionary feature of the temporal enhancement layer, the H.263+ standard stipulates that the pictures included in the temporal scalable mode should be bidirectional predictive (B) pictures as shown in the video stream of FIG. 3 .

Fig. 3 shows a schematic illustration of a scalable video configuration 300, illustrating B-picture prediction dependencies known in the art of video coding. The initial intra-coded frame (I ₁ ) 310 is followed by a bi-directional predictive frame (B ₂ ) 320 . This is followed by a (uni-directional) predictive frame (P ₃ ) 330 and then a second bi-directional predictive frame (B ₄ ) 330 . This is followed by a (unidirectional) predicted frame ( _P5 ) 350, and so on.

Fig. 4 is a schematic illustration of a layered video configuration known in the field of video coding technology. A layered video bitstream includes a base layer 405 and one or more enhancement layers 435 .

The base layer (Layer 1) includes one or more intra-coded pictures (I-pictures) 410 that are sampled, encoded and/or compressed from the original video signal pictures. In addition, the base layer includes a plurality of predicted inter-coded pictures (P-pictures) 420 , 430 predicted from the intra-coded frame 410 .

In an enhancement layer (layer 2 or 3 or more) 435, three types of pictures can be used:

(i) bidirectional prediction (B) picture (not shown);

(ii) an enhanced interpolation (EI) picture 440 based on an intra-coded picture 410 of the base layer 405; and

(iii) Enhancement prediction (EP) pictures 450, 460 based on inter-coded pictures 420, 430 of the base layer.

Vertical arrows going up from lower layers illustrate that pictures in the enhancement layer are predicted from reconstructed approximations of pictures in the reference (lower) layer.

In conclusion, scalable video coding can be used in multipoint multimedia conferencing, and only in the case of point-to-point or multipoint video communications. However, current wireless networks do not support multipoint communication. Alternatively, with multipoint communication, each layer is sent in a separate multipoint communication session, and the receiver determines whether it is logged into one or more sessions.

Therefore, there is a need for an improved method of videoconferencing configuration and operation that mitigates the above-mentioned disadvantages.

Contents of the invention

According to the present invention, there is provided a method for relaying video images in a multimedia video conference as claimed in claim 1, a video conferencing device for relaying video images as claimed in claim 7, a A wireless device for participating in a video conference as claimed in claim 11, a multi-point processor as claimed in claim 12, a video communication system as claimed in claim 16, a system as claimed in claim 18 A media resource functional element, a video communication unit as claimed in claim 19 or claim 20, and a storage medium as claimed in claim 23. Other aspects of the invention are set out in the dependent claims.

In summary, the inventive principle of the present invention is to solve the disadvantages of the prior art configurations by providing a video exchange method to improve identification of participants and speakers in a video conference. The present invention utilizes layered video coding to better utilize the bandwidth available to each user.

Description of drawings

Figure 1 shows a known centralized conference model.

Fig. 2 shows a functional block diagram of a traditional video exchange mechanism.

Fig. 3 is a schematic illustration of a video configuration representing picture prediction dependencies known in the field of video coding technology.

Fig. 4 is a schematic illustration of a layered video configuration known in the field of video coding technology.

Exemplary embodiments of the invention will now be described with reference to the accompanying drawings, in which:

Fig. 5 shows a functional block diagram of a video exchange mechanism according to a preferred embodiment of the present invention.

Fig. 6 shows a functional block diagram/flow diagram of a multi-point processing unit according to a preferred embodiment of the present invention.

Figure 7 shows a video display of a wireless device participating in a video conference using the preferred embodiment of the present invention.

Fig. 8 shows a UMTS (3GPP) communication system employed in accordance with a preferred embodiment of the present invention.

Detailed ways

In general, the preferred embodiments of the present invention propose a new video switching mechanism for multimedia conferencing using layered video coding. Previously, layered video coding was only used to divide a video bitstream into more than one layer: a base layer and one or several enhancement layers as described above with reference to FIG. 4 . These known techniques for scalable video communications are described in detail in standards such as H.263 and MPEG-4.

However, the inventors of the present invention have realized that these benefits can be obtained by adopting the principles of layered video coding and applying the adopted principles to multimedia videoconferencing applications. In this way, the present invention defines a scalable video coding for multimedia conferencing as opposed to peer-to-peer or multipoint video communications.

Referring now to FIG. 5, there is shown a functional block diagram 500 of a video switching mechanism in accordance with a preferred embodiment of the present invention. Compared with traditional centralized conference system formation, this video exchange is performed as follows. MCU 520 , such as MCU 520 located within Internet Protocol (IP) based network 510 , includes switch 530 .

Notably, the MCU 520 receives a 'layered' video stream comprising the base layer 552, 562, 572, 582 and one or more enhancement layer streams of all parties (user equipment) 550, 560, 570, 580 555, 565, 575, 585. For purposes of clarity, only one enhancement layer video stream per participant is shown.

MCU 520 may also receive combined (multiplexed) audio streams 590 from participants individually. The MCU 520 then uses the switch 530 to select the base layer video stream for the number of active speakers 535 and the enhancement layer 540 for the most active speaker. The MCU 520 then sends these video streams 535 , 540 to all participants 550 , 560 , 570 , 580 .

Preferably, the selection process of determining the most active speakers is performed by the MCU 520 analyzing the audio stream 590 to first determine who all of these active speakers are. The most active speaker is then determined, preferably in the multipoint processor unit, as described in FIG. 6 . Preferably, one or more base layers and one enhancement layer are sent to the participants according to a priority based on each participant's activity.

To implement the improved but more complex video switching mechanism of FIG. 5, a multipoint processing unit (MP) 600 is adapted to facilitate a new video switching mechanism according to a preferred embodiment of the present invention and as shown in FIG.

MP 600 also receives audio stream 590 from a participant's video/multimedia communication unit via packet filtering module 610 and routes the audio stream to packet routing module 630 . However, the audio stream is now also routed to a speaker identification module 620 which analyzes the audio stream 590 to determine who is the active speaker. Speaker identification module 620 assigns priorities based on each participant's activity and determines:

(i) most active speakers 620,

(ii) any other active speakers 625 and absent persons

(iii) Any remaining inactive speakers.

According to a preferred embodiment of the present invention, the speaker identification module 620 then forwards the priority information to the switching module 640, which is adapted to handle the speaker's priority. In addition, the switching module 640 is adapted to receive layered video streams from a participant's video communication unit via the packet filtering module 610, the layered video streams comprising video base layer streams 552, 562, 572 and 582 and video enhancement layer streams 555, 565 , 575 and 585. The switching module 640 uses the speaker information to send the video base layer of the second (secondary) active speaker and the most active speaker and the video enhancement layer of the most active speaker to all participants through the packet routing module 630 .

Accordingly, one or more receive ports of the multipoint processor are adapted to receive layered video streams from a set of user equipment 550, 560, 570, and 580, the layered video streams including base layer video streams 552, 562, 572, and 582 and enhancement layer video streams 555, 565, 575 and 585. In contemplation of the present invention, if it is determined that there is only one active speaker, the switching module 640 may select only one base layer video image and the corresponding one or more enhancement layers. That speaker is then automatically designated as the most active speaker for transmission to one or a group of user devices 550, 560, 570, and 580.

As happens in video conferencing, when the most active speaker changes frequently, enhancement layers will be constantly swapped. The inventors of the present invention have recognized potential problems with such frequent and rapid exchanges. In this case, if the first frame is actually a predicted frame (EP) from the previous only second active speaker, then that frame needs to be converted to an interpolated frame (EI).

To address this potential problem, video base layer streams 552 , 562 , 572 and 582 and video enhancement layer streams 555 , 565 , 575 , 585 from packet filtering module 610 are preferably input to unpacketization function 680 . Depacketization function 680 demultiplexes the video stream and provides the demultiplexed video stream to video decoder and buffer function 670 .

To synchronize and coordinate video decoding, the video decoder and buffer function 670 receives an indication of the most active speaker 622 . After extracting the video stream information of the most active speaker, the video decoder and buffer functional element 670 provides bi-directional prediction (BP) 675 and/or prediction (EP) video stream data of the most active speaker 622 to 'EP frame to EI Frame decoding module '660. The 'EP Frame to EI Frame Decoding Module' 660 processes the input video stream to provide the original speaker enhancement layer video stream as intra-coded (EI) frames.

The initial speaker enhancement layer video stream is then input to the packetization function 650 where it is packetized and input to the switching module 640 . The switch module 640 then combines the initial speaker enhancement layer video stream and the second active speaker's video base layer stream 552 , 562 , 572 and 582 and routes the combined multimedia stream to the packet routing module 630 . The packet routing module then routes the information to the participants according to the method of FIG. 5 .

In a preferred embodiment of the present invention, the video exchange module 640 uses the output of the 'EP frame to EI frame decoding module' 660 when determining that the initial speaker change has occurred.

It is contemplated by the present invention that one or more modules similar to module 660 may also be included in MP 600 to perform the same function on the second speaker when they are deemed to have changed. Otherwise, in embodiments where a single 'EP Frame to EI Frame Decoding Module' 660 is used to decode the video stream of the initial speaker, when an inactive speaker is assumed to become the second active speaker, the speaker identification Module 620 (or switch module 640) may request a new interpolated frame. Alternatively, the exchange module 640 may wait for a new interpolated frame of the new second active speaker before sending the corresponding video base layer stream to all participants.

In addition to the preferred embodiment of the present invention, it is also contemplated by the present invention to use multiple types of speakers in cases where more than one enhancement layer can be used. By using multiple classes of speakers, more accurate scalability of multimedia messages can be obtained due to improved speaker recognition, especially for large video conferences.

Adding the conversion of predicted frames to interpolated frames for one or more base layer streams is also contemplated by the present invention. In this way, the switching module 640 can quickly switch between base layers without waiting for new interpolation frames.

Figure 7 shows a video display 710 of a wireless device 700 participating in a video conference using the preferred embodiment of the present invention. By implementing the principles of the invention described previously, improved video communications can be obtained. Specifically, for a given bandwidth, by reducing the video quality of the next (second) active speaker 730 and not providing video for inactive speakers, the participant is now able to receive a better video of the most active speaker 720 quality. To provide this improved video conferencing, the video communication device receives the enhancement and base layers of the most active speaker 720, the base layer of the second active speaker 730, and receives no video from inactive speakers.

In this way, the video communication unit can provide a continuously updated video image of the most active speaker on the larger, higher resolution display, while the smaller display can show the second (secondary) active speaker.

Preferably, the wireless device 700 has a primary video display 710 for displaying a higher quality video image of the most active speaker, and one or more second, different displays for displaying each of the less active speakers. The processing of the respective video images into the respective displays is preferably performed by a processor (not shown) operatively coupled to the video displays. The processor receives indications of the most active speaker 720 and the second active speaker and determines which of the received video images should be displayed on the first display and which image received from the second active speaker 730 should be displayed on the second display show. Advantageously, a second display can be provided to provide a lower quality video image of the second active speaker, thereby saving costs.

It can be expected that in the future, MCU-based systems will facilitate multimedia communications over IP-based networks. Therefore, it is contemplated by the inventors of the present invention that the techniques described herein can be incorporated into any H.323/SIP based multipoint multimedia conference or system utilizing an MCU.

The foregoing preferred application is in the Third Generation Partnership Project (3GPP) specification for the Wideband Code Division Multiple Access (WCDMA) standard. In particular, the invention can be applied in the IP multimedia domain (described in the 3G TS25.xxx series of specifications), which plans to incorporate H.323/SIP MCUs into 3GPP networks. Referring to Figure 8, the MCU will be supported by a Media Resource Function (MRF) 890A.

Figure 8 shows a 3GPP (UMTS) communication system/network 800 in a hierarchical structure, which can be employed in a preferred embodiment according to the present invention. The communication system 800 is adapted to and contains network elements capable of operating over UMTS and/or GPRS air interfaces.

This network is generally considered to include:

(i) User Equipment Domain 810, consisting of:

(a) User SIM (USIM) field 820, and

(b) mobile device domain 830; and

(ii) Infrastructure domain 840, consisting of:

(c) access network domain 850, and

(d) core network domain 860, which consists of (at least) the following:

(di) service domain 870, and

(dii) transit domain 880, and

(diii) IP Multimedia domain 890 with multimedia provided by SIP (ETFRFC2543).

In mobile device domain 830 UE 830A receives data from subscriber SIM 820A in USIM domain 820 via a wired Cu interface. UE 830A communicates data with Node B 850A in network access domain 850 via the wireless Uu interface. Within the Network Access Domain 850, Node B 850A contains one or more transceiver units and communicates with the rest of the cellular based system infrastructure, eg RNC 850B, via the Iub interface defined by the UMTS specification.

RNC 850B communicates with other RNCs (not shown) via the Iu interface. RNC 850B communicates with SGSN 870A in serving network domain 870 via the Iu interface. Within service network domain 870, SGSN 870A communicates with GGSN 870B via Gn interface, and SGSN 870A communicates with VLR server 870C via Gs interface. According to a preferred embodiment of the present invention, SGSN 870A communicates with an MCU (not shown), which is located within the media resource function element (890A) of IP multimedia domain 890 . Communication is performed via the Gi interface.

GGSN 870B (and/or SSGN) is responsible for interfacing UMTS (or GPRS) with a Public Switched Data Network (PDSN) 880A such as the Internet or the Public Switched Telephone Network (PSTN). The SGSN870A performs the routing and tunneling functions for traffic within the UMTS core network, while the GGSN870B connects to the external packet network, in this case any UMTS-mode network that accesses the system.

RNC 850B is a UTRAN element responsible for resource control and allocation of many Node Bs; typically, one RNC 850B can control 50 to 100 Node Bs. RNC850B also provides reliable user business transmission through the air interface. Multiple RNCs communicate with each other (via interface Iur) to support handover and macrodiversity.

SGSN870A is the UMTS core network element responsible for session control and interface to location registers (HLR and VLR). SGSN is a large centralized controller for many RNCs.

GGSN870B is a UMTS core network element, responsible for centralizing and tunneling user data of the core packet network to the final destination (for example, Internet Service Provider (ISP)). Such user data includes multimedia and associated signaling data to/from IP multimedia domain 890 . In the IP multimedia domain 890, the MRF is divided into a multimedia resource function controller (MRFC) 892A and a multimedia resource function processor (MPFP) 891A. As mentioned above, the MRFC892A provides multipoint controller (MC) functionality, while the MPFP891A provides multipoint processor (MP) functionality.

The protocol used across the Mr reference point/interface 893A is SIP (as defined in RFC2543). Call State Control Function (CSCF) 895A acts as a call server and handles multimedia call signaling.

Therefore, according to a preferred embodiment of the present invention, elements SGSN 870A, GGSN 870B and all parts of MRF 890A are adapted to facilitate multimedia messages as described heretofore. Furthermore, UE 830A, Node B 850A, and RNC 850B are also adapted to facilitate improved multimedia messaging, as described hereinbefore.

In general, this adaptation can be carried out in the respective communication unit in any suitable way. For example, new devices may be added to an existing communication unit, or alternatively existing portions of an existing communication unit may be employed, such as by reprogramming one or more processors therein. Thus, the required adaptations may be implemented in the form of processor-implementable instructions stored on a storage medium such as a floppy disk, hard disk, PROM, RAM, or any combination of these or other stored multimedia.

Alternatively, such adaptation of the multimedia message may also be controlled, fully or partially implemented by employing any other part of the communication system 800, which is also contemplated by the present invention.

While the above elements are typically provided as discrete units (on their own respective software/hardware platforms), divided into mobile device domain 830, access network domain 850, and service network domain 870, it is contemplated that other configurations may also be employed.

Also, in the case of other network infrastructures, such as a GSM network, implementation of the processing operations may be performed by any suitable node, such as any other suitable type of base station, base station controller, mobile switching center or operational and management control device and so on. Alternatively, the above-mentioned steps may be performed by various components distributed at various locations or entities within any suitable network network.

As mentioned above, preferably, when applied in centralized video conferencing, the video conferencing method using layered video coding can provide the following advantages:

(i) Speaker recognition is much improved compared to conventional systems, since the shared bandwidth allows sending one or more enhancement layers and a few base layers instead of just one full-quality video stream.

(ii) Video switching using the inventive principles described here is smoother when the active speaker changes because it defines several states, active speaker, second most active speaker, inactive speaker.

(iii) The video quality of the most active speakers has been improved.

(iv) The improved video communication unit can display various speakers, each displayed image depending on the priority associated with the transmission of the corresponding video communication unit.

A method of relaying video images in a multimedia video conference between a plurality of multimedia user equipment has been described. The method comprises the steps of: transmitting a layered video image comprising a base layer and one or more enhancement layers through a plurality of a plurality of user equipments, and receiving the transmitted layered video image at a multipoint control unit. A number of base layer images of many active speakers and one or more enhancement layers of the most active speakers are selected. The multipoint control unit transmits a plurality of base layer video images of a plurality of active speakers and one or more enhancement layers of a most active speaker to one or more of a plurality of multimedia user devices.

Furthermore, a video conferencing apparatus for relaying video images between a plurality of user equipment is described. Additionally, a wireless device for participating in a video conference in which a number of participants send video images is described.

Claims (23)

1. A method for relaying video images in a multimedia conference between a plurality of multimedia user equipment (550, 560, 570, 580), the method comprising the steps of: transmitting a layered video image by a plurality of said groups of user equipment, wherein said layered video image includes a base layer (552, 562, 572, 582) and one or more enhancement layers (555, 565, 575, 585); Receive the layered video images sent at the multipoint control unit (520); selecting a number of base layer video images of a number of active speakers (535) and one or more enhancement layers (540) of the most active speakers; and sending, by the multipoint control unit (520), a plurality of base layer video images of the plurality of active speakers (535) and one or more enhancement layers (540) of the most active speakers to the plurality of multimedia user devices (550 , 560, 570, 580) one or more. 2. The method of relaying video images in a multimedia conference according to claim 1, wherein the selecting step further comprises the step of: A number of audio data streams (590) transmitted by said plurality of multimedia user devices (550, 560, 570, 580) are analyzed to determine said number of active speakers and/or a most active speaker. 3. The method for relaying video images in a multimedia conference according to claim 1 or 2, wherein the method further comprises the steps of: assigning a priority to each layered video image and/or said audio data stream sent by a respective user equipment; and Based on the assigned priorities, a number of base layer video images (535) and one or more enhancement layers (540) are selected for transmission to all of the plurality of multimedia user equipments (550, 560, 570, 580) one or more of the above. 4. A method of relaying video images in a multimedia conference according to any preceding claim, wherein the method further comprises the step of: The first predicted frame of the video image of the most active speaker is decoded (660) as an intra-coded frame for enhancing the video quality of the most active speaker. 5. A method of relaying video images in a multimedia conference according to any preceding claim, wherein the method further comprises the step of: When more than one enhancement layer is available, an indication of the classification of the one or more speakers is received by the multimedia control unit (520) along with the transmission of each layered video image, so as to provide a classification of the video image More precise measurability. 6. A method of relaying video images in a multimedia conference according to any preceding claim, wherein the method further comprises the step of: For one or more base layer video streams, convert predicted frames to intra-coded frames. 7. A video conferencing device for relaying video images between a plurality of user equipment (550, 560, 570, 580), the video conferencing device comprising: A multipoint control unit (520) adapted to receive a plurality of layered video images of said plurality of user equipment transmissions, wherein said layered video images include a base layer (552, 562, 572, 582) and one or more enhancement layers (555, 565, 575, 585); and a video exchange module (530), operatively coupled to said multipoint control unit (520) and adapted to select a number of base layer video images of a number of active speakers (535) and one or more of the most active speakers enhancement layer (540); wherein The multipoint control unit (520) is further adapted to send a plurality of base layer video images of the plurality of active speakers (535) and one or more enhancement layers (540) of the most active speakers to the plurality of multimedia user devices One or more of (550, 560, 570, 580). 8. The video conferencing device of claim 7, further comprising: a predicted frame to intra-coded frame decoding module (660), operatively coupled to said video exchange module (530), such that if said frame is initially received by said multipoint control unit (520) as a predicted frame, it Provides the most active speaker enhancement layer video stream as intra-coded frames. 9. A video conferencing device according to claim 7 or 8, further comprising: A speaker identification module (620) analyzes a number of audio streams (590) to determine a number of active speakers and/or said most active speaker. 10. The video conferencing device of claim 9, wherein said speaker identification module (620) assigns priorities based on the determined activity of each participant to determine one or more of the following: most active speaker (622 ), any other active speakers (625), and any inactive speakers. 11. A wireless device (700) for participating in a video conference, in which a plurality of participants send video images, the wireless device (700) comprising: a video display (710) having a first display and one or more second, different displays for displaying respective parties (720, 730) from the plurality of parties; and a processor, operatively coupled to said video display, for receiving an indication of a most active speaker (720) and a plurality of next active speakers (730), and determining to receive from said most active speaker (720) The video images are displayed on the first display providing higher quality video images, and the video images received from the plurality of secondary active speakers (730) are displayed on all the lower quality video images on the second display. 12. A multipoint processor comprising: One or more receiving ports adapted to receive a layered video image comprising a base layer video stream (552, 562, 572, 582) and an enhancement layer from a plurality of user equipments (550, 560, 570, 580) video streams (555, 565, 575, 585); and a switching module (640), operatively coupled to the one or more receive ports, selects a number of base layer video images of a number of active speakers (535) and one or more enhancement layers (540) of the most active speakers, for transmission to one or more user equipments (550, 560, 570, 580). 13. The multipoint processor according to claim 12, further comprising: a speaker identification module (620), operatively coupled to said one or more receive ports, for analyzing a plurality of audio streams (590) received from a plurality of said plurality of user devices to determine a plurality of active speakers and/or the most active speakers described. 14. The multipoint processor according to claim 12 or 13, wherein said speaker identification module (620) assigns priorities based on the determined activity of a number of participants to determine one or more of the following: most active speaker (622), any other active speakers (625), and any inactive speakers. 15. A multipoint processor according to any one of claims 12 to 14, further comprising: a predicted frame-to-intra-coded frame decoding module (660), operatively coupled to said switching module (640), such that if said most active speaker's enhancement layer video stream received at a corresponding port is a predicted frame, Just convert it to an intra-coded frame. 16. A video conferencing device adapted to perform the method steps of any one of claims 1 to 6, or adapted to include any one of claims 7 to 10, or adapted to include a plurality of point processor for video communication systems. 17. The video communication system according to claim 16, wherein the video communication system is compatible with the UMTS communication standard (800) with the Internet Protocol Multimedia Domain (890) to facilitate video conference communication. 18. A video conferencing device adapted to perform the method steps of any one of claims 1 to 6, or adapted to include any one of claims 7 to 10, or adapted to include any one of claims 12 to 15 The media resource function element (890A) of the point processor. 19. A video communication unit (700) adapted to receive a layered video conference image produced according to the method of claims 1 to 6. 20. A video communication unit adapted to generate a layered video conference image for use in the method of claims 1 to 6, or to transmit a layered video conference image generated according to the method of claims 1 to 6. 21. The video communication unit of claim 19, wherein the video communication unit is one of: Node B (850A), RNC (850B), SGSN (870A), GGSN (870B), MRF (890A). 22. A method of relaying video images in a multimedia video conference according to claims 1 to 6 or a video conferencing device according to any one of claims 7 to 10, or a multipoint processor according to any one of claims 12 to 15 Or a video communication system according to claims 16 to 17, or a media resource functional element (890A) according to claim 18 or a video communication unit according to claims 19, 20 or 21, all adapted to facilitate Standard video conferencing image. 23. A storage medium storing processor implementable instructions for controlling a processor to perform the method of any one of claims 1 to 6.

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