CN101180822A - Enhanced voip media flow quality by adapting speech encoding based on selected modulation and coding scheme (mcs) - Google Patents

Abstract

A voice-over-IP (VoIP) connection is established over a radio interface with a mobile radio station. A current radio condition for the VoIP connection is determined, and from that, a modulation and coding scheme (MCS) for a portion of the VoIP connection. A VoIP voice encoding mode for a portion of the VoIP connection is selected based on the determined modulation coding scheme. VoIP voice is then encoded into a number of VoIP encoded frames using the selected VoIP voice encoding mode which has an associated bit rate. An optimal number of VoIP encoded frames is included in a VoIP packet for transport over the VoIP connection given the selected voice encoding mode and the selected MCS. Other adjustments may be made to ensure robustness of the VoIP connection and/or to maximize capacity.

Description

By adapting to enhanced voip media flow quality based on the speech coding of selected Modulation and Coding Scheme (MCS)

Technical field

The present invention relates to telecommunications and found to be applied to the favourable example that IP phone (VoIP) is communicated by letter.

Background technology

VoIP is to use the voice service of Internet protocol (IP) to transmit.In the world that moves, VoIP means that using packet switching (PS) business to be used to transmit Internet protocol (IP) divides into groups, and this IP grouping comprises adaptive multi-rate (AMR) the codec speech frame that for example is used for the speech mobile calls.The packet switching connection abbreviates data usually as and connects.

Packet switching network uses circuit switching to be used to carry voice service, and wherein Internet resources were assigned to receiver from sender statically before message transmits beginning, has therefore created in " circuit ".During whole message transmitted, the resource reservation was exclusively used in this circuit, and whole message is along same path.Get fairly good though this arrangement is used for transmitting speech work, it is an attractive selection that but IP transmits for speech, reason wherein is a lot, comprises relatively low equipment cost, speech and comprises comprehensive, relatively low bandwidth requirement that the data of multimedia (as Email, short message, video, World Wide Web (WWW) or the like) are used and the wide usability of IP.

In packet switching network, message is broken down into grouping, and each grouping can take different routes to arrive the destination, and grouping is recompiled into origination message in the destination.Packet switching (PS) business that is used for VoIP can be for example GPRS (general packet radio service), EDGE (enhanced data rates global evolution) or WCDMA (Wideband Code Division Multiple Access (WCDMA)).Each of these professional examples all is based upon on the global system for mobile communications (GSM) just, promptly originally is the second generation (2G) the digital radio access technology of Europe exploitation.GSM is enhanced in 2.5G, to comprise the technology such as GPRS.The third generation (3G) comprises the mobile phone technology that is covered by the IMT-2000 of International Telecommunications Union (ITU) system.Third generation partner program (3GPP) be for standardization IMT-2000 based on the part of WCDMA and a group the International Organization for Stand, operator and the distributors of working.

EDGE (GPRS (EGPRS) that is called enhancement mode sometimes) is the 3G technology that the data rate of image width band is provided to mobile device.EDGE allows the consumer to be connected to the internet and transmits and receive data, and comprises digital picture, webpage and photo, may be also fast than 3 times of speed using general GSM/GPRS network.EDGE makes the GSM carrier can provide the mobile data of higher rate to insert, and serves more mobile data client, and discharges the GSM network capacity to hold extra voice service.EDGE uses the carrier bandwidths of TDMA (time division multiple access) frame structure, logic channel and the 200kHz identical with the GSM network, and it allows the design of existing sub-district to keep intact.

In the EDGE technology, base station transceiver (BTS) communicates with travelling carriage (for example, cell phone, portable terminal or the like comprise the computer such as kneetop computer with portable terminal).Base station transceiver (BTS) has a plurality of transceivers (TRX) usually.As time division multiple access (TDMA) radio communications system of GSM, GPRS and EDGE time and space is divided into time slot on the particular radio-frequency.Time slot is by component frame, and wherein the user is assigned with one or more time slots.In packet switching TDMA, even a user may be assigned with one or more time slots, but other user may use same time slot.Therefore, need time slots scheduler (scheduler) to guarantee that time slot suitably and is effectively distributed.

EDGE provides nine kinds of different Modulation and Coding Scheme (MCS): MCS1 to MCS9.Relatively low encoding scheme (for example, but MCS1-MCS2) provide more reliable slower bit rate and be suitable for relatively poor radio condition.Higher relatively encoding scheme (for example, MCS8-MCS9) is transmitted higher bit rate, but is required better radio condition.Which MCS link-quality control (LQC) selects to use based on current wireless electricity situation under each particular case.

In EDGE, LQC is radio link control (RLC) data block selection MCS of each Temporary Block Flow (TBF).TBF is that the logic between travelling carriage (MS) and the packet control unit (PCU) connects.PCU usually (but not necessarily) for example be arranged in radio access network at base station controller (BSC).TBF is used for up link or down link transmits the GPRS grouped data.Carrying out actual grouping on physical data radio channel (PDCH) transmits.Therefore effectively select the bit rate of TBF by selecting MCS, and changed its bit rate by the MCS that change is used for TBF.

Advanced many speed (AMR) speech frame comprises voice, is generally 20 milliseconds of voice by the AMR codec encodes.It is the compressed digital form that voice encryption device, vocoder and codec are used interchangeably and refer to voice/speech coding.The AMR codec supports different bit errors to detect and prevention (UED/UEP).By bit being classified as more responsive and more insensitive kind in the perception, UEP/UED mechanism allows voice to carry out more effective transmission on the network diminishing.Have only when in the most responsive bit, finding one or more bit errors, declare that just frame is damaged and do not transmitted.On the other hand, have one or more bit errors if transmit speech frame in more insensitive bit, then the voice quality based on the human auditory still is considered to and can accepts.The key property of the high bit-error as EDGE (BER) environment is to have robustness by redundant and bit error and sensitiveness classification for the packet loss that is provided by the AMR codec.

Another benefit of AMR is that the adaptation rate that (on-the-fly) takes over seamlessly between codec mode when being used for the free time is adaptive.Speech quality along with changing the special speed of ratio and therefore obtaining can use a large amount of AMR codec modes.The AMR codec can comprise a plurality of narrowband codec patterns: 12.2,10.2,7.95,7.4,6.7,5.9,5.5 and 4.75kbit/s.Or even the broadband of 12.65kbit/s (WB) Mode A MR WB also is available.

Usually, connect for VoIP, the end points of VoIP communication is for example called out travelling carriage A and called mobile station B, consults which AMR codec mode and will be used for the VoIP connection.If travelling carriage A represents that it can use AMR codec mode 1,2 and 3, default mode is an AMR codec mode 2, if and B represents that it can use AMR codec mode 2,3 and 4, default mode is an AMR codec mode 2, may will select AMR codec mode 2 so.Usually make the initial selected of AMR codec mode then based on the bit rate of wanting that is used to communicate by letter at application protocol layer.As a result, the codec mode that is used for voip call is chosen in and makes on the application layer and need not to know current wireless radio channel condition or selected MCS.Determine the current wireless radio channel condition and select MCS for the transmission of next radio data piece, all promptly on rlc/mac layer, carry out at relatively low radio access protocol layer.

Because EDGE is by selecting MCS to change the bit rate that is used for TBF according to the radio condition on each particular radio block gap, so this bit rate variation is very fast.As a result, the static state of VoIP AMR encoder or codec mode selects to cause usually being lower than optimum performance, for example lower than required speech quality speech quality.For example, if selected maximum bit rate, high speech quality encoder or codec mode then may generate data sometimes on the bit rate higher than current air transfer rate permission, cause VoIP to be grouped in receiving terminal broadcast time and arrive too late after having passed through.Another problem that the static state of VoIP AMR encoder or codec mode is selected is, when current wireless electricity situation is fairly good, if selected VoIP encoder or codec mode are low bit speed rate, low speech quality encoder, the data that then send in radio blocks are wanted much less than the data that sent.In other words, can receive much better speech quality at one of receiving terminal, and need not extra bandwidth cost, but really not so because of the utilization of resources of difference.

Relevant issues are hardware and bandwidth usages of poor efficiency.For the higher relatively bit rate of using EDGE to provide is provided, each radio blocks that is used for specific MCS encoder should be filled full as far as possible.For example, the MCS-8 radio blocks can be held 1088 bits.If encoder only has 500 bits to send, utilized so to be less than 50% possible EDGE throughput, it is converted into relatively low bit rate.

A kind of solution of these problems may be pattern or the codec mode that changes voice encryption device on radio interface according to the overall data throughput of measuring.But this method is not suitable for " carrier (bearer) ", and as EDGE TBF, it is along with the radio condition that changes changes each radio blocks.In other words, even the user consults specific bit speed when setting up TBF, the actual bit speed on that TBF changes according to fast-changing current wireless electricity situation.Therefore, when being received at the network entity place that can change the voice encryption device pattern to measured total throughout, fast-changing radio condition has made that throughput value out-of-date.

Summary of the invention

The inventor has imagined quality and the capacity that better method solves these problems and improves IP phone (VoIP).Setting up VoIP by radio interface and travelling carriage is connected.Determine the current wireless electricity situation that VoIP connects, and be used for the Modulation and Coding Scheme (MCS) that a part of VoIP connects from its selection.The VoIP speech coding or the encoding/decoding mode that are used for the VoIP connection of a part are determined based on selected Modulation and Coding Scheme.The VoIP speech uses the selected VoIP speech coding pattern with related bits speed to be coded as a plurality of VoIP coded frame then.The optimal number of VoIP coded frame is included in and is used in the VoIP grouping transmitting wherein given selected speech coding pattern and selected MCS by the VoIP connection.The quantity of VoIP coded frame is variable and can changes along with the change of MCS.

Can make other and adjust the total capacity that guarantees robustness and/or the more performance that VoIP connects or increase the communication system of supporting VoIP connection and other connections.For improving robustness, on the lower data rate of the data rate that can support than determined MCS, select VoIP speech coding pattern, and/or can select lower MCS than the current wireless electricity determined MCS of situation.If radio communications system is the system of time division multiple access (TDMA) type, as GPRS and EDGE, a more time slot of the number of timeslots that then determined transmission VoIP grouping is required also can be used to improve robustness.

Can make other and adjust the capacity that increases the VoIP connection.For example, the VoIP grouping of being created is used to form one or more radio blocks and is used for transmitting by radio interface.The quantity that is included in the VoIP coded frame in the VoIP grouping has a mind to be selected to " filling " wireless radio transmission piece, wherein given selected speech coding pattern and selected MCS.

This method is dynamic rather than static.By the information about the carry-on actual voice amount of bits of every radio blocks under given current wireless electricity situation and selected MCS situation is provided to the VoIP voice encryption device in application layer, the speech coding parameter that voice encryption device can just will use is made wiser decision.Detect the variation of the radio condition that is used for the VoIP connection, and can make one or more variations in response to this.When radio condition worsens, can carry out one or more the following steps: reduce MCS, reduce the speed of VoIP voice encryption device, increase number of timeslots, and/or be the quantity that the VoIP voice encryption device frame that every IP of radio blocks divides into groups is filled in selected MCS and selected VoIP voice coder rate adjustment.On the other hand, when radio condition is improved, one or more following steps can be performed: increase MCS, increase the speed of VoIP voice encryption device, reduce the quantity of time slot, and/or be the quantity that the VoIP voice encryption device frame that every IP of radio blocks divides into groups is filled in selected MCS and selected VoIP voice coder rate adjustment.

Description of drawings

Fig. 1 is the simplification functional block diagram of exemplary mobile radio communicaltions system;

Fig. 2 is the communication protocol figure of EDGE (enhanced data rates global evolution) system;

Fig. 2 connects the exemplary non-limiting step of coming radio resource allocated and carrying out that be determined symmetry or asymmetric or the flow chart of action by the radio resources allocation controller based on the data of being asked;

Fig. 3 is the simplification functional block diagram of travelling carriage, shows according to a non-restrictive illustrative up link execution mode mutual between voice over ip feature of carrying out on the different agreement layer and protocol layer;

Fig. 4 is the mutual simplification functional block diagram between the voice over ip feature of carrying out on the different nodes that illustrates according to a non-restrictive illustrative down link execution mode, and described node comprises IMS node, BSC node and BTS node;

Fig. 5 is the block diagram of the exemplary implementation detail of explanation in packet control unit (PCU);

Fig. 6 is the block diagram of the exemplary implementation detail of explanation in BTS;

Fig. 7 is the block diagram of the exemplary implementation detail of explanation in the IMS node; And

Fig. 8 is the block diagram of the exemplary implementation detail of explanation in travelling carriage;

Fig. 9 is explanation AMR 4.75NB codec transmits a plurality of time slots of VoIP data with every IP grouping 2AMR frame on different C/I figure; And

Figure 10 is that explanation AMR 12.65WB codec transmits the figure of a plurality of time slots of VoIP data with every IP 2 AMR frames that divide into groups on different C/I.

Embodiment

In the following description, in order to explain and non-limiting purpose, set forth detail, so that complete understanding of the present invention is provided such as ad hoc structure, interface, technology etc.Yet, it will be apparent to those skilled in the art that the present invention can realize in deviating from other embodiment of these details.That is, those skilled in the art can design various arrangements, although these arrangements less than clearly describing or showing, have embodied principle of the present invention here and have been included within its aim and the scope.In some instances, the detailed description of known device, circuit and method is omitted to avoid with the fuzzy description of the invention of unnecessary details.Here put down in writing all statements of principle of the present invention, aspect and embodiment and the equivalent that concrete example is intended to comprise its 26S Proteasome Structure and Function thereof.In addition, such equivalent is intended to comprise the current known equivalent and the equivalent of exploitation in the future, promptly developed no matter what structure but carry out any element of identical function.

Therefore, for example, it should be appreciated by those skilled in the art that the block diagram here can represent to embody the concept map of the illustrative circuit of know-why.Equally, be to be understood that the various processes that expressions such as any flow chart, state transition graph, false code can be represented on computer-readable medium in fact and therefore carried out by computer or processor, no matter whether such computer or processor is clearly demonstrated.

The function of various elements comprises the functional block that is noted as " processor " or " controller ", and these functions can provide by using specialized hardware and the suitable software of combination of hardware that can executive software.When function was provided by processor, this function can be by single application specific processor, provide by single shared processing device or by a plurality of separate processors, and wherein some can be shared or distribute.And, clearly the using of term " processor " or " controller " should not be considered the hardware that special finger can executive software, and can comprise but be unlimited to digital signal processor (DSP) hardware, be used for read-only memory (ROM), random-access memory (ram) and the nonvolatile memory of storing software.

Fig. 1 shows exemplary mobile radio communications system 10, this system is coupled to one or more circuit-switched networks 12 such as PSTN (PSTN) and/or integrated services digital network (ISDN) etc. by mobile switching centre (MSC) 16 core network nodes, and is coupled to one or more packet switching networks 14 as the internet by Serving GPRS Support Node (SGSN) 20 and Gateway GPRS Support Node (GGSN) 22.PSTN 12 and ISDN 14 are circuit switched core network and MSC core network node 16 support circuit switched service.The internet is a packet-switched core network, and SGSN 20 and GGSN 22 are packet-switched core network nodes.Except these core networks and relevant core network node, also have internet protocol multimedia subsystem (IMS) 13, it provides IP-based business, as VoIP and multimedia service.IMS 13 can comprise that media resource function (MRF) 15 is to transmit the business based on medium.IMS is coupled to core network, GGSN 22 and SGSN 20.MSC 16, IMS 13 and SGSN 20 are coupled to mobile subscriber database, for example home subscriber server (HSS) 18 and be coupled to radio access network.

In this non-limitative example, Radio Access Network is based on GSM's and is known as base station system (BSS) 24.Here at this radio access network that can be applied to other types based on the technology described in the GSM/EDGE system.BSS 24 comprises the one or more base station controllers (BSC) 26 (only showing) that are coupled to a plurality of base station transceivers (BTS) 28.Base station controller 26 is controlled the radio resource and the dedicated radio link of the sub-district of being served by BTS 28 under its control.BTS 28 and mobile radio unit (MS) 30 use radio communication to communicate by air interface.The one or more sub-districts of each base station transceiver (BTS) 28 services.For each sub-district of serving, base station transceiver 28 provides radio transmission resources pond (managed by BSC usually and distribute) to be used for communicating with the travelling carriage of that sub-district.Each base station (BTS) 28 comprises that controller and radio set and baseband processing circuitry come to handle wireless radio transmission and reception in each sub-district of serving.

Each travelling carriage (MS) 30 comprises that radio set and data processing and controlled entity/function are used to provide IP phone (VoIP) ability.Person of skill in the art will appreciate that travelling carriage 30 and its data processing and control generally include many other function and applications.Travelling carriage 30 also comprises input-output apparatus, such as display screen, keypad, loud speaker, microphone or the like.

In EDGE, EGPRS or GPRS, the first link layer protocol context, be called Temporary Block Flow (TBF), be based upon on the up link from the travelling carriage to the radio net, and the 2nd TBF is based upon on the down link from the radio net to the mobile radio unit.Travelling carriage (MS) and the logic between the packet control unit (PCU) that TBF can be considered in the network connect.Though PCU can be arranged in BSC 26, PCU can also be arranged in BTS 28, be arranged in SGSN 20 or the like.Fig. 2 is the communication protocol figure of EDGE appreciated by those skilled in the art system.TBF is shown in the interim connection between radio link control (RLC) protocol layer entity among BSC and the MS.Be connected in case up link TBF has been established for data with down link TBF, radio resource (time slot in the EDGE type system) can be assigned to support the connection on the radio/air interface so.Base station controller (BSC) the 26 LLC frame (in Fig. 2, being described as be in " switching " on the BSS) of between travelling carriage (MS) 30 and core network, transferring.The data block that media interviews control (MAC) layer-management is caused by various TBF multiplexing, described TBF is effective available physical radio channel, arbitrate by timeslot scheduling mechanism between various mobile subscribers, this mechanism is organized among the BSC, wherein selects TBF for each time slot.

Usually, PCU carries out LQC and can be arranged in BSC, BTS, SGSN etc.Unrestricted in order only to be easy to describe, suppose that PCU is in BSC.BSC 26 selects Modulation and Coding Scheme MCS to be used for the VoIP transmission of each 20 milliseconds of (msec) wireless radio transmission piece in this non-limiting example.The good wireless electricity situation that is used for the VoIP transmission means that more VoIP coded-bit can be included in each 20 milliseconds of radio blocks; Therefore, select higher Modulation and Coding Scheme (MCS).Following form shows the VoIP coded-bit of each 20 milliseconds of radio blocks of each Modulation and Coding Scheme (MCS) that is used for EDGE.

Encoding scheme	MCS1	MCS2	MCS3	MCS4	MCS5	MCS6	MCS7	MCS8	MCS9
										Bit/radio blocks (20ms)	176	224	296	352	448	592	896	1088	1184
Bit rate (kbps)	9.5	11.2	14.8	17.6	22.4	29.6	44.8	54.4	59.2

Form 1

Usually the supposition available bandwidth decides speech coding or codec mode based on the speech quality of hope, and does not consider to be used for the current wireless electricity situation that VoIP connects.The inventor recognizes that better VoIP communication can speech be provided selected MCS to obtain by the VoIP application layer of speech coding, framing and encapsulation by not only giving the channel encoder that is used for chnnel coding and modulation radio piece and modulator but also give wherein before wireless radio transmission.

For this reason, Fig. 3 shows the simplification functional block diagram of travelling carriage (MS) 30, wherein locate being used to of selecting related to the MCS of the VoIP communication of MS 30, not only offer the conventional channel coding that is used for radio blocks and the low protocol layer of EGPRS of modulation but also the VoIP that offers on the higher application protocol layer and use at reference marker (1).Locate at reference marker (2), use that information, VoIP is applied as voice coder/decoder (codec) pattern that VoIP communication selects to be suitable for current wireless electricity situation.In EGPRS, codec is the AMR codec.For example select MCS based on current detected radio condition by link-quality controller (LQC) 32 shown in Fig. 5.Higher codec mode is corresponding to higher bit rate codec output, and lower codec mode is corresponding to lower bit rate codec output.

Locate at reference marker (3), the codec of travelling carriage selects to be used for many AMR frames of IP grouping.Given selected AMR codec mode, the quantity of AMR frame is optimised, and to fill the radio blocks size, it will be used for the selected MCS on the EGPRS layer.For example, 20 milliseconds voice can be coded as an AMR frame, and it is consistent with 20 milliseconds of radio blocks.AMR frame packed (be packaged into IP grouping) and locate IP at reference marker (4) then and be grouped in and be sent to the EGPRS layer in the travelling carriage 30, this layer correspondence SNDCP/LLC and rlc/mac layer as shown in Figure 2.By data volume that relatively needs to send and the data volume that is fit to each time slot, EGPRS layer formation wireless radio transmission piece carries grouped data and locates at reference marker (5), selects a plurality of time slots to carry each wireless radio transmission piece.For example, if only need 1 time slot and 1 time slot to distribute to MS by PCU, MS sends this data so.If only need 1 time slot and 2 time slots to distribute to MS by PCU, MS sends these data and request release TBF so.If desired 2 time slots and only 1 time slot distribute to MS by PCU, MS begins to send these data and request PCU is upgraded to 2 time slots so.2 time slots and 2 time slots are distributed to MS by PCU if desired, and MS sends this data so.Shown in reference marker (6), the wireless radio transmission piece is sent out by chnnel coding and modulation and by radio interface according to the selected MCS that is used for this a part of VoIP transmission.

On the down link that VoIP connects, carry out similar program, but this function is carried out by different entities or node preferably.About this point, Fig. 4 shows the simplification functional block diagram of IMS node 14, wherein is provided for the VoIP that carries out uses in the IMS node at reference marker (1) the selected MCS that locates to communicate by letter for VoIP.Locate at reference marker (2), use that information, VoIP application choice AMR codec mode is used for VoIP communication, and this pattern is suitable for current wireless electricity situation.Locate at reference marker (3), selected AMR codec mode selects to be used for a plurality of AMR frames of IP grouping.Given selected AMR codec mode, the quantity of AMR frame is optimized to fill the radio blocks size, and it will be used for the selected MCS on the EGPRS layer.The AMR frame is packed, and locates the EGPRS layer that the IP grouping is sent to use packet control unit 31 realizations as shown in Figure 5 at reference marker (4) then.In this example, CPU is arranged in BSC 26.Packet control unit 31 forms the wireless radio transmission pieces and carries grouped data, and the time slots scheduler 40 relevant with packet control unit 31 locates to select a plurality of time slots to carry each wireless radio transmission piece for example can be similar to above-described mode about travelling carriage at reference marker (5).This wireless radio transmission piece is provided for one or more base stations 28, physical layer operations is carried out in base station 28, this physical layer operations comprises according to being this part VoIP transmission (one or more grouping) selected MCS chnnel coding and modulation radio transmission block, and during selected time slot, transmit modulation intelligence, shown in reference marker (6) by air interface.

Fig. 5 shows link-quality controller (LQC) 32 with the simplified block diagram form, and it is included in this example in the packet control unit (PCU) 31.Moreover PCU 31 can be arranged in BSC, base station or such as the core network node of SGSN.LQC 32 comprises MCS selector 34, and MCS selector 34 comprises that MCS selects enquiry form 36.The input of selecting form 36 can be the radio condition of one or more that detect and VoIP join dependencys, such as RSSI, SIR, CIR, BER, BLER or the like.Good wireless electricity situation causes higher quantity (but higher throughput relatively poor robustness), and MCS is selected, and relatively poor radio condition causes (but lower throughput better robustness) MCS of lower quantity selected.Selected MCS is provided for time slots scheduler 40, and scheduler program 40 also receives the VoIP grouping from IMS node 14.Time slots scheduler 40 becomes the wireless radio transmission piece with the VoIP packet switched, and its size is decided based on selected MCS.Use the program of for example being explained as the step among Fig. 3 (5), time slots scheduler 40 determines to carry a plurality of time slots of radio blocks at selected MCS.Radio blocks that forms among the PCU31 and selected time slot are forwarded to suitable base station 28 and are used for being transferred to travelling carriage 30 by air interface.

Fig. 6 shows the simplified block diagram of exemplary base 28.TBF data queue 70 buffer memory wireless radio transmission pieces are used for downlink transmission to travelling carriage 30.Use to be the selected MCS of this TBF, the radio blocks that ejects from formation 70 channel encoder 72 by chnnel coding and modulated in modulator 74.Modulation output is sent out by radio in RF transmitter 76 then.The information that RF transmitter 76 receives about time slot from time slots scheduler 40 transmits the modulation radio blocks of data during this time slot.The uplink radio block that receives from travelling carriage 30 is also transmitted to BSC 26 in the base station, but RF reception, demodulation and the channel-decoding piece that is used for uplink communication is not shown.Base station 28 detect the signal quality of the uplink communication that receives from travelling carriage 30 and give LQC 32 provide the radio condition information that is detected be used to determine/adjust selected MCS.In one exemplary embodiment, upgrade radio condition information for each 20 milliseconds of radio blocks.

Fig. 7 shows the exemplary reduced block diagram of IMS node, and wherein this IMS node can use MRF entity 15 to realize.Receive selected MCS or determine selected MCS from the information that provides by the MCS selector 34 the BSC in addition.Based on selected MCS, codec mode selector 80 is selected relevant codec mode, and this pattern has the related bits speed of the voice that are used for speech coding.Selected codec mode is provided for AMR codec 82, and it is according to selected codec mode speech coding VoIP voice.Codec framer 84 receive the speech data of speech codings and according to selected AMR codec with that data framing, selected AMR codec itself is based on current MCS and selects and select.Codec framer 84 generates a plurality of frames to be included in by VoIP wrapper 86 in the VoIP grouping.That quantity decides based on selected codec mode, so that fill the wireless radio transmission piece best, the wireless radio transmission piece will be used for the EGPRS layer based on selected MCS.In other words, codec framer 84 can select the AMR codec frames of correct number to fill the radio blocks size of being managed by selected MCS, is also managed by selected MCS because be provided for the selected AMR codec mode of codec framer 84.The VoIP grouping is provided for the base station by BSC.

With reference to Fig. 8, travelling carriage 30 is carried out the similar functions that is used for up link, because up link can have different MCS with down link.Travelling carriage comprises AMR codec mode selector 50, and its reception is used for the selected up link MCS of this TBF.Up link MCS is determined by the MCS selector among the BSC 34 and is sent to MS, as shown in piece 51.Selected codec mode is provided for AMR codec 52, the VoIP voice that its speech coding received.Output bit flow in AMR codec framer 54 according to selected codec mode by framing, as above explain for down link.The AMR codec frames is formed in the VoIP grouping, and this grouping is stored in the TBF data queue 58 then.According to selected MCS, grouping is by chnnel coding and modulated.Modulating data is formed in the wireless radio transmission piece, and this transmission block is transmitted by the time slot of being discerned by time slots scheduler 40.

Consider such example, wherein the VoIP application choice VoIP codec mode in travelling carriage or IMS node is a VoIP speech coding bit with 20 milliseconds of VoIP speech codings.Higher codec mode means better speech quality, because more bits is carried this VoIP speech of 20 milliseconds.Following form 2 has comprised the different AMR codec modes or the EDGE example of speed.

AMR codec mode/speed	4.75	7.95	12.65WB	(12.65WB every IP grouping 2AMR frame)
					Each has the bit number of the AMR speech frame (20ms) of header overhead	224	288	376	640

Form 2

Following form 3 shows minimum MCS, its time slot that can be used and be still varying number (for example, 0.5,1,1.5 or 2) fill the IP grouping with the speech frame of two speech codings, wherein said time slot is used to transmit the 20 millisecond radio blocks relevant with that grouping.If use the time slot of lesser amt, must use higher MCS so, with the so much data of abundant transmission with higher bit rates.By increasing more time slot, can use more low bit speed rate, the MCS of robustness more.Therefore form 2 shows and needs how many time slots to be used for given MCS pattern.Just enough for MCS 1 and 4.75, one time slots of AMR, therefore there is no need to use 1.5 or 2 time slots, because the single time slot that sends in that 20 milliseconds of radio blocks interims for all data adaptings of 20 ms intervals.May need more time slot for AMR WB 12.65.Therefore, if radio condition worsens, require lower MCS, then this time slots scheduler can increase employed number of timeslots, so that keep the bit rate of being transmitted by selected codec mode.Otherwise, when radio condition worsens, will need to reduce codec rate, so that successfully transmit the VoIP data.

The AMR pattern	Employed time slot
					0.5	1	1.5	2
	320 bits in AMR 4.75:2 encapsulated frame/IP grouping  IP grouping	MCS	5	MCS 1					-	-

640 bits in AMR WB 12.65:2 encapsulated frame/IP grouping  IP grouping

MCS 7

MCS 5

MCS 2

MCS 1

Form 3

Following form 4 shows the average packet size that is used for two different AMR codec modes, i.e. AMR 4.75 (arrowband (NB)) and AMR 12.65 (broadband (WB)).Codec is included in more frame in the IP grouping, this grouping size increase but cost is a bit rate reduces.The maximum quantity of the AMR frame that every IP that the quantity of the frame of every IP grouping can be held based on the MCS block size divides into groups is selected as up to the configurable limit of maximum.So this weighs between the capacity utilization of time of transmitter side buffer memory voice and radio net.

Audio coder ﹠ decoder (codec)	AMR 4.75(NB)			AMR-WB 12.65(WB)
							The frame number of every IP grouping	1	2	3	1	2	3
Speech data *[bit]	112	208	312	264	528	787
							Average IP header **[bit]	32	32	32	32	32	32
LLC+SNDCP[bit]	80	80	80	80	80	80
							Average packet size	224	320	424	376	640	896
Mean bitrate [kbit/s]	11.2	8	7.1	18.8	16	14.9

Form 4

Lack the coordination between low protocol layer and the selection of VoIP codec mode, VoIP connects and to be subjected to adverse influence-or because the voice delivery of poor efficiency or because low-quality voice delivery, wherein low protocol layer be each radio blocks transmission process MCS selection, and the VoIP codec mode is chosen on the higher application protocol layer and makes.For example, suppose that the VoIP encoder selects low quality/low bit speed rate codec mode based on historical events: the voice that are 20 milliseconds produce 224 bits.Suppose that the MCS selector is these 20 milliseconds based on current wireless electricity situation and selects high MCS-7, have 897 bits to use like this.Do not know the capacity that this is higher if VoIP uses, then only use 25% available 897 bits.Listener can be experienced better speech quality and not need extra bandwidth cost.

In order to reach better effect, can use one of several configurable selections.For example, the VoIP encoder is apprised of the availability of MCS-7 transmission and is therefore become AMR12.65WB from AMR 4.75.In the sort of situation, be that 20 milliseconds of speech intervals produce 376 bits rather than 224 bits.These speech bits are sent in a time slot immediately, 1 AMR frame of every IP grouping.As a result, receiver receives better speech quality, and extra " cost " of the system that need not.Alternatively, the VoIP encoder can be apprised of the availability of MCS-7 transmission and therefore become AMR 12.65WB from AMR4.75.In this case, be that 20 milliseconds of speech intervals produce 376 bits.The AMR codec also becomes 2 AMR frames is packaged into each IP grouping.As a result, produce 640 bits in 40ms speech interim.The voice of that 40ms are sent out by a radio blocks (for example, 1 time slot of a 20ms) then.Compare with first kind of situation, power system capacity doubles, because voice only are to be sent out (the additional buffered time that less cost is 20ms) at interval every a radio blocks on that time slot.

Consider another kind of problematic situation, wherein, based on the historical events of per 20 milliseconds of voice 376 bits of correspondence, the VoIP encoder is selected the codec of bit rate, 12.65WB.On the other hand, the MCS selector has been selected minimum MCS-1 based on the current radio condition of the difference of 176 bits of only can transmitting.But because VoIP in using voice encryption device and do not know that MCS restriction, so the IP grouping arrives packet control unit, have 376 bits.Even this packet control unit can adapt to and distribute two time slots to be used for this connection, but that only provides the capacity of 352 bits, it is still less than required 376.As a result, transmission lags behind the speed that produces data, and the result causes buffer under run operation (under-run) and listener to have lower voice quality.

Can handle this problematic situation better by using technical method as described herein.For example, the AMR codec is apprised of MCS-1 and is selected, and changes into AMR 4.75,2 AMR frames of every IP packet encapsulation, and therefore every 40ms generates 320 bits.The voice of these 40ms can be sent out at interval by two radio blocks then, and wherein each radio blocks carries 176 bits at interval, i.e. 2 * 176=352＞320.As a result, voice continue to flow and are not subjected to the interruption of the travelling carriage of speaker.

Therefore between current MCS and voice codec pattern, exist important mutual.Because codec is provided with selected MCS, so it can make appropriate mode/rate adaptation.The quantity of have a mind to selecting to be included in the VoIP coded frame in the VoIP grouping is filled the wireless radio transmission piece, wherein given selected speech coding pattern and selected MCS.

Can also make other and adjust the total capacity that guarantees robustness and/or the more performance that VoIP connects or increase the communication system of supporting VoIP connection and other connections.In order to improve robustness, VoIP speech coding pattern can be selected on the lower data rate of the data rate that can support than determined MCS, and/or MCS can be selected be lower than the determined current wireless of MCS electricity situation.If radio communications system is the system of time division multiple access (TDMA) type, as GPRS and EDGE, in the then determined number of timeslots that need to transmit the VoIP grouping one more multi-slot can also be used to improve robustness.

Fig. 9 and 10 shows two different examples, and it has illustrated the correlation between radio condition (C/I is a unit with dB), MCS and employed number of timeslots.Fig. 9 is the 4.75AMR pattern of per minute group 2AMR frame, and Figure 10 is the 12.65AMR pattern of per minute group 2AMR frame.Be shown in dotted line the MCS that on each C/I, can select.Solid line shows the AMR number of frames of that specific MCS pattern of needing how many time slots to be used for codec and every IP grouping.These voice are sent on a time slot immediately.As a result, receiver obtains better speech quality, and need not the extra cost of system.

Therefore can make a plurality of adjustment and increase capacity and/or the reliability that VoIP connects.When radio condition worsens, can carry out one or more following steps: reduce MCS, reduce the speed of VoIP voice encryption device, increase timeslot number, and/or be the quantity that the VoIP voice encryption device frame that every IP of radio blocks divides into groups is filled in selected MCS and selected VoIP voice coder rate adjustment.On the other hand, when radio condition is improved, can carry out one or more following steps: increase MCS, increase the speed of VoIP voice encryption device, reduce the quantity of time slot, and/or be the quantity that the VoIP voice encryption device frame that every IP of radio blocks divides into groups is filled in selected MCS and selected VoIP voice coder rate adjustment.Can use other adjustment.

Although specifically illustrated and described various embodiment, claim is not limited to any specific embodiment or example.For example, can use any codec.The example of optional codec comprises: use the MPC-MLQ algorithm G.729, G.729a, G.723.1, use the ACELP algorithm G.723.1, G.711, the iLBC that strengthens of iLBC, RCU, G.729 or G.723.1, strengthen G.711, iPCM-wb, iSAC or the like.Foregoing description is not appreciated that hint any particular element, step, scope or function are essential to make it must be included in the claim scope.The subject area of applying for a patent is only limited by claim.The scope that protects by law is limited by the word of being put down in writing in the claim that is allowed and its equivalent.Should be appreciated that the present invention is not limited to the disclosed embodiments, on the contrary, the present invention is intended to cover the arrangement of various modifications and equivalence.

Claims (35)

1. one kind is used for the method that is connected by the IP phone (VoIP) that radio interface and mobile radio unit (30) are set up, comprises and determining and the electric situation of the current wireless of VoIP join dependency, it is characterized in that:

Based on determined current wireless electricity situation, determine Modulation and Coding Scheme (MCS) for connecting a part of VoIP information that transmits by VoIP;

To small part based on determined Modulation and Coding Scheme, select to be used for the VoIP speech coding pattern that a part of VoIP connects, wherein VoIP speech coding pattern has relevant bit rate;

Use selected VoIP speech coding pattern that the VoIP speech coding is become the VoIP coded frame of variable number, wherein this variable number of VoIP coded frame depends on determined MCS; And

Be included in the quantity that is used for connecting the VoIP coded frame of the VoIP grouping that transmits by VoIP.

2. according to the process of claim 1 wherein that the described variable number of VoIP coded frame is the optimal number that maximization is included in the VoIP bit quantity in the radio blocks that transmits by radio interface.

3. according to the method for claim 1, further comprise:

When determining different MCS, change the variable number of VoIP coded frame for the VoIP connection.

4. according to the method for claim 1, further comprise:

On the lower data rate of the data rate that can support than determined MCS, select VoIP speech coding pattern.

5. according to the method for claim 1, further comprise:

Select to be lower than the MCS of determined MCS for current wireless electricity situation.

6. according to the method for claim 1, wherein this method is used for GPRS or EDGE type system, be used to support that the radio channel resource that connects comprises time slot, and the VoIP speech coding is carried out by adaptive multi-rate (AMR) encoder/decoder (codec) (82).

7. according to the method for claim 6, further comprise:

Determine that a plurality of time slots are used to transmit the VoIP grouping.

8. according to the method for claim 7, further comprise:

Use a more multi-slot, it determines to transmit the quantity of VoIP grouping.

9. according to the method for claim 7, further comprise:

Use the VoIP grouping to form the wireless radio transmission piece;

Use determined MCS chnnel coding and modulation radio transmission block; And

Use the wireless radio transmission piece of the time slot of institute's quantification by radio interface transfer channel coding and modulation.

10. according to the method for claim 9, wherein the wireless radio transmission piece is a fixed size, and this method further comprises:

Adjustment is included in the quantity of the VoIP coded frame in the VoIP grouping of filling the wireless radio transmission piece.

11. the method according to claim 9 further comprises:

Select one or more following steps to increase the robustness that VoIP connects: to reduce MCS, reduce the speed of VoIP voice encryption device, increase number of timeslots, or be the quantity that selected MCS and selected VoIP speech coding mode adjustment are filled the VoIP voice encryption device frame that every IP of radio blocks divides into groups.

12. the method according to claim 9 further comprises:

Be the variation in the VoIP joint detection radio condition; And

If changing is improved radio condition, then select one or more following steps: increase MCS, increase the speed of VoIP speech coding, reduce the quantity of time slot, or be the quantity that selected MCS and selected VoIP speech coding mode adjustment are filled the VoIP voice encryption device frame that every IP of radio blocks divides into groups.

13., be implemented in travelling carriage or internet protocol multimedia subsystem (IMS) node (15) according to the method for claim 1.

14. one kind is used for the equipment that IP phone (VoIP) that mobile node (30) supports to set up by radio interface via radio access network (24) connects, comprises:

Modulation and Coding Scheme (MCS) selector (51) is used for, it is characterized in that for connecting a part of VoIP speech information selective channel encoding scheme and the modulation scheme that transmits by VoIP based on current wireless electricity situation determined and the VoIP join dependency:

VoIP voice encryption device (52), be configured to (1) and select the speech coding pattern based on selected MCS for this part of V oIP speech information at least in part, wherein VoIP speech coding pattern has relevant bit rate, and (2) using selected VoIP speech coding pattern that the VoIP speech information is encoded becomes the VoIP coded frame of variable number, and wherein the variable number of VoIP coded frame depends on selected MCS; And

VoIP wrapper (56) is used for being included in the quantity that is used for connecting by VoIP the VoIP coded frame of the VoIP grouping that transmits.

15. according to the equipment of claim 14, wherein the described variable number of VoIP coded frame is that maximization is included in by the optimal number of mobile node by the VoIP bit quantity in the radio blocks of radio interface transmission.

16. according to the equipment of claim 14, wherein when determining different MCS for the VoIP connection, the VoIP voice encryption device further is configured to change the variable number of VoIP coded frame.

17. according to the equipment of claim 14, wherein the VoIP voice encryption device further is configured to select VoIP speech coding pattern, this pattern has the low bit rate of bit rate that can support than determined MCS.

18. according to the equipment of claim 14, wherein the MCS selector is configured to select be lower than the MCS for the determined MCS of current wireless electricity situation.

19. equipment according to claim 14, wherein this equipment is configured to be used for the GPRS type system, the radio channel resource that wherein is used to support VoIP to connect comprises time slot, and the VoIP voice encryption device is adaptive multi-rate (AMR) encoder/decoder (codec).

20. the equipment according to claim 19 further comprises:

Circuit (64) is used for determining that a plurality of time slots are to be used to transmit the VoIP grouping.

21. the equipment according to claim 20 further comprises:

Circuit (54) is configured to use the VoIP grouping to form the wireless radio transmission piece;

Channel encoder (60) is used for based on selected channel coding schemes chnnel coding information;

Modulator (62) is used for based on the chnnel coding information of selected modulation scheme modulation from channel encoder; And

Radio transceiver circuit (64) is configured to use than the time slot of institute's quantification and Duos one time slot, transmits modulation intelligence by radio interface.

22. according to the equipment of claim 21, wherein the wireless radio transmission piece is a fixed size, and

Wherein the VoIP voice encryption device is configured to adjust the quantity in order to the VoIP coded frame of filling the wireless radio transmission piece that is included in the VoIP grouping.

23. the equipment according to claim 21 further comprises:

Control circuit (51), be configured to select one or more following steps to increase the robustness that VoIP connects: to reduce MCS, reduce the speed of VoIP voice encryption device, increase number of timeslots, or be the quantity that the VoIP voice encryption device frame that every IP of radio blocks divides into groups is filled in selected MCS and selected VoIP voice coder rate adjustment.

24. the equipment according to claim 21 further comprises:

Control circuit (51), be configured to when current wireless electricity situation is improved, select one or more following steps: increase MCS, increase the speed of VoIP voice encryption device, reduce the quantity of time slot, or be the quantity that the VoIP voice encryption device frame that every IP of radio blocks divides into groups is filled in selected MCS and selected VoIP voice coder rate adjustment.

25. one kind is used for the equipment that IP phone (VoIP) that network node (15) supports to set up by radio interface via radio access network (24) connects, it is characterized in that:

Codec mode selector (80), be configured to connect the selected Modulation and Coding Scheme (MCS) that transmits a part of VoIP speech information based on being used for by VoIP to small part, selection is by the speech coding pattern of the VoIP speech information of this part of VoIP connection transmission, and this selected MCS is based on current wireless electricity situation determined and the VoIP join dependency;

VoIP codec (82), being configured to becomes coding VoIP data according to selected speech coding pattern with VoIP speech information coding, and selected VoIP speech coding pattern has relevant bit rate;

Codec framer (84), be configured to use selected VoIP speech coding pattern with the VoIP data framing of coding to produce the VoIP coded frame of variable number, wherein the variable number of VoIP coded frame depends on selected MCS; And

VoIP wrapper (86) is used for being included in the quantity that is used for connecting by VoIP the VoIP coded frame of the VoIP grouping that transmits.

26. according to the network node of claim 25, wherein network node is internet protocol multimedia subsystem (IMS) node (15), it in use is coupled to radio access network.

27. according to the network node of claim 25, wherein selected MCS is by being determined from the information that radio access network received and can changing along with the radio condition that VoIP connects and change.

28. according to the network node of claim 25, wherein the described variable number of VoIP coded frame is that maximization is included in by the optimal number of mobile node by the VoIP bit quantity in the radio blocks of radio interface transmission.

29. according to the network node of claim 25, wherein when determining different MCS for the VoIP connection, the codec framer further is configured to change the variable number of VoIP coded frame.

30. network node according to claim 25, wherein the codec mode selector further is configured to select VoIP speech coding pattern, and this pattern has the lower bit rate of bit rate that can support than determined modulation scheme and determined encoding scheme.

31. according to the network node of claim 25, wherein this network node is configured to be used for the GPRS type network, the radio channel resource that wherein is used to support VoIP to connect comprises time slot, and codec is adaptive multi-rate (AMR) codec (82).

32. the network node according to claim 25; Wherein the codec mode selector is configured to: (1) selects one or more following steps to increase the robustness that VoIP connects: reduce the speed of VoIP voice encryption device or fill the quantity of VoIP voice encryption device frame of every IP grouping of radio blocks for selected MCS and selected VoIP voice coder rate adjustment, perhaps (2) select one or more following steps when current wireless electricity situation is improved: increase the speed of VoIP voice encryption device or fill the quantity of VoIP voice encryption device frame of every IP grouping of radio blocks for selected MCS and selected VoIP voice coder rate adjustment.

33. a radio access node that uses with the network node in the claim 31 comprises:

Link-quality controller (32) is used for determining and the electric situation of the current wireless of VoIP join dependency, and

Packeting controller (40), the VoIP grouping and the formation radio blocks that are used to receive from the VoIP wrapper are used for transmitting by radio interface,

Wherein the codec framer is configured to adjust the quantity that is included in the VoIP coded frame that is used for filling each wireless radio transmission piece in the VoIP grouping.

34. the radio access node according to claim 33 further comprises:

Time slots scheduler (40), it is configured to be identified for to transmit the time slot of the required minimum number of each radio blocks and scheduling and Duos one time slot than the determined quantity that is used to transmit each radio blocks.

35. the radio access node according to claim 33 further comprises:

Control circuit (31), be configured to: (1) selects one or more following steps to increase the robustness that VoIP connects: reduce MCS or increase number of timeslots, and (2) select one or more following steps when current wireless electricity situation is improved: increase MCS or fill the quantity of VoIP voice encryption device frame of every IP grouping of radio blocks for selected MCS and selected VoIP voice coder rate adjustment.

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