WO2008098490A1

WO2008098490A1 - Method and apparatus for adjusting audio codecs

Info

Publication number: WO2008098490A1
Application number: PCT/CN2008/070026
Authority: WO
Inventors: Hailei Wang; Shijun Li; Haopeng Zhu; Feng Li; Yong Wang; Xinhua Yang
Original assignee: Huawei Technologies Co., Ltd.
Priority date: 2007-02-12
Filing date: 2008-01-04
Publication date: 2008-08-21
Also published as: CN101056466B; CN101056466A

Abstract

The present invention provides a method for adjusting audio codecs. After a call originated from a user terminal is received, the priority list of the audio codecs is adjusted according to the self-supported audio codecs reported by the user terminal, in order that the audio codec occupying less resources has a higher priority; and the audio codec used by the user terminal is determined according to the adjusted priority list of the audio codecs. In addition, the present invention provides an apparatus and Mobile Switch Center for adjusting audio codecs. Since the audio version used by the user terminal occupies less network resources than the original audio version, the technical schemes provided by the present invention can save network resources, and furthest utilize network resources. In addition, the method and apparatus provided by the present invention can not only save the bearing-network resources, but also save the air interface resources.

Description

TECHNICAL FIELD The present invention relates to a wireless communication technology, and more particularly to a method and apparatus for adjusting a voice codec mode. Background of the invention

With the separation of call control and bearer operations in the core network and the progress of IP over the entire network, the resource occupancy in the bearer network, especially the bandwidth occupancy in the network, has become a major concern of the entire core network.

Most of the current user terminals support multiple voice versions, such as full rate version 1, full rate version 2, full rate version 3, half rate version 1, and so on. As shown in Table 1, the correspondence between several voice versions and their corresponding codes and rates is listed.

Voice version Codec rate (kBit/s)

GSM speech full rate version 1 GSM FR 13.0

GSM speech full rate version 2 GSM EFR 12.2

GSM speech full rate version 3 FR AMR 4.75-12.2

GSM speech full rate version 4 OFR AMR-WB 6.6-23.85

GSM speech full rate version 5 FR AMR-WB 6.6-23.85

GSM speech half rate version 1 GSM HR 5.60

GSM speech half rate version 3 HR AMR 4.75-7.95

GSM speech half rate version 4 OHR AMR-WB 6.6-23.85

GSM speech half rate version 6 OHR AMR 6.6-23.85

UMTS Adaptive Multi-Rate UMTS — AMR 4.75-12.2

UMTS Adaptive Multi-Rate 2 UMTS_AMR2 4.75-12.2 8PSK Half Rate Adaptive Multi-Rate OHR — AMR 4.75-12.2

Full Rate Adaptive Multi-Rate WideBand FR AMR-WB 6.6-12.65

UMTS Adaptive Multi-Rate WideBand UMTS AMR-WB 6.6-23.85

8PSK Full Rate Adaptive Multi-Rate

OFR AMR-WB 6.6-23.85 WideBand

8PSK Half Rate Adaptive Multi-Rate

OHR AMR-WB 6.6-12.65 WideBand

According to the existing protocol standards, the core network selects the voice version with the highest priority according to the voice version reported by the user terminal. Generally speaking, in order to ensure the quality of the call of the user terminal, the highest priority voice version usually requires a large network resource. For example, a list of voice versions supported by the user terminal: GSM speech full rate version 1 , GSM speech full rate version 2 , GSM speech full rate version 3 and GSM speech half rate version 1 (1), where full rate version 1 is the highest priority voice version. In this case, according to Table 1, the full-rate version 1 requires more network resources than the full-rate version 2, full-rate version 3, and half-rate voice version 1 to occupy network resources.

Here, the voice version is a name in a conventional GSM network, and the voice version of the softswitch network defined in the current 3GPP is called a voice codec mode. In this paper, the voice version and the voice codec represent the same concept.

In the prior art, the priority order of each voice codec mode reported by the user terminal to the voice codec mode list of the core network is determined by the user terminal, and is fixed; meanwhile, according to the provisions of the existing protocol standard, the network side The voice encoding and decoding mode with the highest priority among the user terminal voice codec mode list is always taken. Here, the user terminal can support multiple voice encoding and decoding modes, but the network side does not support the voice encoding and decoding according to the user terminal. In the manner of saving network resources, the voice codec mode used by the user terminal during the call is selected. Therefore, due to improper selection of voice codec mode, network resources are wasted and network resources cannot be utilized to the greatest extent. Summary of the invention

In view of this, the embodiments of the present invention provide a method and an apparatus for adjusting a voice codec mode. The method and apparatus provided by the present invention can save network resources and utilize network resources to the greatest extent.

In addition, the present invention also provides a mobile switching center, which can save network resources and utilize network resources to the greatest extent.

To achieve the above objective, the technical solution of the embodiment of the present invention is implemented as follows: A method for adjusting a voice codec mode, the method includes the following steps: after receiving a call initiated by a user terminal, according to a voice codec reported by the user terminal In the manner of adjusting the priority of the voice codec mode, the voice codec mode with low resource occupation has a high priority;

The voice codec mode used by the user terminal is determined according to the priority of the adjusted voice codec mode.

A device for adjusting a voice codec mode, the device comprising:

An adjustment unit, configured to adjust a priority of the voice codec mode, so that a voice codec mode with a low resource occupation has a high priority;

The execution unit is configured to determine, according to the priority of the voice codec mode adjusted by the adjustment unit, that the user terminal uses the highest priority voice codec mode.

A mobile switching center, the mobile switching center at least comprising: an adjusting unit, an executing unit, and an intersecting unit;

The adjusting unit is configured to adjust a priority of a voice codec mode supported by the user terminal The priority of the voice codec mode with low resource occupation is high;

The intersection unit is configured to: cross the voice codec mode supported by the base station controller or the wireless network terminal controller, and the voice codec mode supported by the user terminal adjusted by the adjustment unit; and obtain the intersection The list is sent to the execution unit;

The executing unit is configured to determine, according to the received list, a voice encoding and decoding mode in which the user terminal uses the highest priority.

The method for adjusting the voice codec mode provided by the embodiment of the present invention, the priority of the voice codec mode supported by the user terminal is adjusted to ensure the priority of the voice codec mode with low resource occupation is high; The voice codec mode priority is used to determine the voice codec mode used by the user terminal. In addition, the present invention also provides an apparatus for adjusting a voice codec mode. At the same time, the present invention also provides a mobile switching center. The technical solution provided by the present invention adjusts the priority of the voice codec mode supported by the user terminal, so that the highest priority voice codec mode occupies resources, and thus the voice codec mode used by the user terminal is compared with the previous one. The voice codec method used occupies less network resources, thus saving network resources and maximizing the use of network resources. In addition, the method and the device provided by the present invention can not only save the bearer network resources, but also save the air interface resources, and can widely apply the traditional TDMA exchange GSM network, or the bearer and control separation but the A interface has no IP network and bearer and control. Separate the A-interface IP-based softswitch system. BRIEF DESCRIPTION OF THE DRAWINGS

Figure 1 is an exemplary flow chart of the method of the present invention;

Figure 2 is an exemplary structural view of the apparatus of the present invention;

3 is a flow chart of a method according to a first preferred embodiment of the present invention;

4 is a structural diagram of a mobile switching center according to a first preferred embodiment of the present invention; Figure 5 is a flow chart of a method of a second preferred embodiment of the present invention;

6 is a structural diagram of a mobile switching center according to a second preferred embodiment of the present invention. Mode for carrying out the invention

The present invention will be further described in detail below with reference to the accompanying drawings.

The core idea of the embodiment of the present invention is: adjusting the priority of the voice codec mode supported by the user terminal, so that the voice codec mode with low resource occupation has a higher priority; and determining the user terminal according to the adjusted voice codec mode priority. The voice codec method used. Since the network side selects the highest priority voice codec mode in the user terminal, it is used as the voice codec mode used by the user terminal. Therefore, in the voice codec mode supported by the user terminal, the priority of the resource occupation is lowered, and the network side selects the voice codec mode with high priority but low resource occupation according to the priority level.

Currently, the priority of each voice version is determined based on its location in the voice version list in the user terminal. For example, when the user terminal reports the voice version list of the core network in the order of full rate version 1, full rate version 2, full rate voice version 3, and half rate voice version 1, the full rate version 1 has the highest priority. , followed by full rate version 2, full rate speech version 3, and half rate speech version 1. Therefore, this method of determining the priority based on the location can also be referred to as the priority order. In the embodiment of the present invention, the voice version priority supported by the user terminal can be adjusted by adjusting the position of the voice version in the voice version list in the user terminal.

Referring to Figure 1, Figure 1 is an exemplary flow chart of the method of the present invention. The method includes: in step 101, the mobile switching center adjusts the priority of the voice codec mode according to the resource optimization principle according to the voice codec mode supported by the user terminal; in step 102, the mobile switching center according to the mobile switching center The priority of the adjusted voice codec mode is determined The terminal uses the highest priority voice codec.

Referring to Figure 2, Figure 2 is an exemplary structural view of the apparatus of the present invention. The apparatus includes: an adjustment unit 21 and an execution unit 22. The adjusting unit 21 is mainly used to adjust the priority of the voice codec mode supported by the user terminal according to the resource optimal principle. The execution unit 22 is mainly used to determine the priority of the voice codec mode adjusted by the adjustment unit 21, and determine the highest priority voice codec mode for the user terminal.

Here, the resource optimization principle used in the embodiment of the present invention is: the priority of the voice codec mode with low resource occupation is high. The specific method may be that the voice codec mode with the smallest resource occupation is given the highest priority, the resource occupancy second is ranked second, and so on; the multi-rate adaptive voice codec mode may also be ranked as the highest priority. The voice codec mode with the smallest occupied bandwidth ranks second, and the second most occupied bandwidth is ranked second. The most occupied bandwidth is ranked last. It can also be based on the resources used by the network. When the network resources are still sufficient, the resources will be used. The voice encoding and decoding method, which is not the smallest, gives the highest priority.

In addition, in the embodiment of the present invention, in addition to adjusting the priority of the voice coding and decoding mode according to the resource optimal principle, the priority of the user terminal user may also be considered at the same time. For example, for a user terminal with a lower priority than the original user, the voice codec mode with the smallest resource occupation is given the highest priority; and for the user terminal with a higher user priority, the user terminal with the higher priority can be lower than the original occupied resource. The voice codec mode gives the highest priority, and differentiated services for user terminal users are realized.

In the embodiment of the present invention, when the current bearer network does not implement the IP address of the A interface, the voice codec mode support capability of the base station controller (BSC) where the call is located may be matched with the voice codec mode supported by the user terminal. Therefore, the voice codec mode used by the user terminal is determined, so that the voice codec mode used by the user terminal matches the version supported by the BSC, thereby achieving better call quality.

In the case that the current bearer network has realized the IP of the A interface, in order to implement the band Transcoder Free Operation (TrFO), to achieve better voice quality, can further determine the voice codec mode used by the user terminal according to the voice codec mode supported by the calling/called user terminal in the call. . Here, the voice codec mode supported by the master/called user terminal is: the intersection of the list of voice codec modes supported by the calling side and the list of voice codec modes supported by the called party on the called side. .

The following describes the first preferred embodiment of the present invention by taking the priority of adjusting the voice codec mode of the user terminal and determining the voice codec mode used by the user terminal during the call.

Referring to Figure 3, Figure 3 is a flow chart of a method in accordance with a first preferred embodiment of the present invention. The use of the flow shown in Figure 3 is the Global System for Mobile Communications (GSM).

In step 301, during the call setup process, the mobile switching center (MSC) in the core network according to the received voice codec mode list supported by the user terminal, and the network resources and resources required by each voice codec mode are the most The principle of excellence is to adjust the priority of the voice codec mode. Here, the referred network resource may be bandwidth.

For example, the priority order of the user terminal supporting the voice codec mode is full rate voice version 1, full rate voice version 2, full rate voice version 3, and half rate voice version 1. When the resource optimization principle is adopted, the multi-rate adaptive voice codec mode is ranked first, the voice codec mode with the smallest bandwidth is ranked second, and the occupied bandwidth is ranked second, and the bandwidth is the largest. At the end, the priority of the adjusted user terminal supporting the voice codec mode may be: full rate voice version 3, half rate voice version 1, full rate voice version 2, full rate voice version 1.

In step 302, the MSC performs an intersection of the priority of the voice codec mode supported by the user terminal and the voice codec mode supported by the BSC of the current call, and obtains a voice codec mode supported by both the user terminal and the BSC. This step may be performed after step 301, or may be performed before step 301. The voice codec modes supported by the BSC where the user terminal is located are: full rate voice version 1, full rate voice version 2, full rate voice version 3. According to the example in step 301, it can be concluded that the voice codec mode that can be used on the air interface of the call is: full rate voice version 3, full rate voice version 2, full rate voice version 1.

In step 303, since the A interface is IP-based in the MSC, in order to implement the TrFO, the MSC can also take the intersection according to the voice codec mode supported by the MGW, that is, the CODEC, and obtain the voice supported by the user terminal, the BSC, and the MGW. Codec mode.

When the GSM CODEC supported by the MGW to which the call is located is GSM FR, FR AMR, GSM EFR and HR AMR, the result according to step 302 and the correspondence between the voice version shown in Table 1 and Codec are obtained. Relationship, the order of the CODEC that this call can use on the MGW is FR AMR, GSM EFR, GSM FR, which corresponds to full rate voice version 3, full rate voice version 2, full rate voice version 1.

In step 304, the MSC selects the voice codec mode with the highest priority as the voice codec mode used by the user terminal according to the intersection obtained in step 303.

In the call setup process, the process shown in FIG. 3 may be performed only for the calling user terminal, or may be performed only for the called user terminal, or may be used for both the primary and the called. If only used for the calling party, the core network can obtain the voice codec mode supported by the user terminal from the call setup (SETUP) message sent by the calling party when the call is initiated. If it is only used for the called party, the core network can obtain the voice codec mode supported by the called user terminal from the call confirmation (CALL CONFIRM) message sent by the user terminal.

If the example of the present invention is used for both the primary and the called, the CODEC negotiation may be performed between the primary and the called party according to the execution result of the step 303, and the calling and the called party support the CODEC. Then, in the CODEC supported by the calling party and the called party, according to the priority of the corresponding voice codec mode, the minimum voice codec mode of the occupied network resource used by the calling party and the called party is determined. In addition, the example of the present invention may also perform the process shown in FIG. 3 after the main called party respectively determines the priority of the voice codec mode used by the calling party, and then intersect the voice codec mode supported by the calling party and the called user to obtain the master. The voice codec mode supported by the called user terminal and occupying the smallest resource. For example, the priority of the voice codec mode determined by the calling party after performing step 303 is: full rate voice version 3, full rate voice version 2, full rate voice version 1; correspondingly, called: full rate voice version 2. Full-rate voice version 1, then the voice version used by the primary and called user terminals is full-rate voice version 2.

The method of the first preferred embodiment of the present invention can be applied not only to the call setup process but also to the handover procedure. For example, the user terminal switches from the 3G UMTS to the 2G GSM handover process. In the handover process, when the MSC of the core network receives the reconfiguration request (Iu-Relocation-Required) reported by the user terminal from the Iu-interface, the MSC switches the priority of the voice codec mode supported by the user terminal to the user terminal to The speech codec mode supported by the target BSC is used as an intersection, and the speech codec mode supported by both the user terminal and the BSC is obtained. Here, a method for obtaining a speech codec mode supported by both the user terminal and the target BSC can be referred to in step 302. The voice codec mode supported by the user terminal is a voice codec mode after the priority is adjusted during the call setup process.

Of course, the technical solution of the present invention may also be: adjusting the voice codec mode supported by the user terminal during the call setup process, and determining the voice version used during the call. When the user terminal switches, the same reference is made. The method introduced in step 302 re-determines the voice codec mode used after the user terminal switches. Similarly, the user terminal may also adjust the voice codec mode supported by the user during the call setup process, and perform a method for determining the voice codec mode used by the user terminal only during the handover process. When the intra-office handover occurs in the user terminal, the MSC described herein is the MSC where the user terminal is currently located; when the inter-office handover occurs in the user terminal, the MSC described herein is the target MSC to which the user terminal is handed over. In the introduction of this article, the MSCs in these two cases are collectively referred to as the target MSC. In the introduction of the first preferred embodiment, the process of adjusting the voice codec mode supported by the user terminal is performed by the MSC each time the user terminal establishes a call or performs handover. In a specific implementation, the adjusted voice codec mode may also be saved, and the voice codec mode used by the user terminal is determined according to the saved voice codec mode in a subsequent process. Here, the operation of storing the adjusted speech codec mode may be performed by a visit location register (VLR), or an HLR, or an MSC. The specific operation is: before the MSC adjusts the voice codec mode of the user terminal, it is determined whether the adjusted voice codec mode of the user terminal is stored in the VLR, or the HLR, or the MSC, and if stored, directly according to the save The voice codec mode corresponding to the current user terminal determines that the user terminal uses the voice codec mode with the highest priority; otherwise, performs the step of adjusting the voice codec mode of the user terminal.

Referring to Figure 4, Figure 4 is a block diagram of a mobile switching center suitable for use in the first preferred embodiment of the present invention. The MSC includes at least: an adjustment unit 41 and an execution unit 42. The adjusting unit 41 is mainly configured to adjust the priority of the received voice codec mode supported by the user terminal according to the resource optimal principle. The executing unit 42 is configured to determine, according to the priority of the voice codec mode adjusted by the adjusting unit 41, that the user terminal uses the highest priority voice codec mode.

The mobile switching center further includes: an intersection unit 43, in order to implement an intersection of the voice codec modes supported between the user terminal and the BSC. The intersection unit 43 is mainly configured to take the voice codec mode supported by the base station controller or the radio network controller, and the voice codec mode supported by the user terminal adjusted by the adjustment unit 41, and send the list obtained by the intersection to the execution. Unit 42. Correspondingly, the execution unit 42 determines the voice encoding and decoding method according to the received list.

In addition, the MSC further includes: a control unit 44 and a storage unit 45. The storage unit 45 is located between the adjustment unit 41 and the execution unit 42 for receiving and saving the tone. The whole unit 41 adjusts the result of the voice decoding mode of the user terminal. The control unit 44 is configured to receive a voice encoding and decoding method supported by the user terminal, and determine whether the corresponding adjustment result is saved in the storage unit 45. When the storage unit 45 saves the corresponding adjustment result, the instruction execution unit 42 is configured according to the storage unit. As a result of the adjustment in 45, the operation of determining the speech codec mode used by the user terminal is performed; when the storage unit 45 does not save the corresponding adjustment result, the adjustment unit 41 is instructed to perform the operation of the adjustment.

Referring to Figure 5, Figure 5 is a flow chart of a method in accordance with a second preferred embodiment of the present invention. The preferred embodiment describes that when the A interface is IP-enabled, the calling user switches the 2G GSM and the relay MGW from the 3G UMTS. Since the A interface is IP-based, in order to implement TrFO, it is necessary to negotiate a common speech codec mode between the calling/called user terminals. The specific steps are as follows:

In step 501, the target MSC intersects the priority of the voice codec mode supported by the user terminal with the voice codec mode supported by the target BSC to which the user terminal is switched, and the voice codec mode supported by the target MGW, to obtain the user terminal and The speech codec mode supported by both the target BSC and the target MGW, such as (V, W, X, Y, Ζ). The voice encoding and decoding method with the highest priority is configured to be used on the target MGW and the target BSC, that is, the voice codec mode V.

Here (V, W, X, Υ, Ζ) respectively represent five kinds of speech codec modes, and the priority of V to Ζ is sequentially lowered. The voice codec mode supported by the user terminal may be a voice codec mode after the priority is adjusted during the call setup process.

Here, the target BSC to which the calling user terminal and the calling user terminal are switched; and the list of voice codec modes supported by the target MGW to which the calling user terminal is switched are referred to as voice editing supported by the calling side. The list of decoding modes may also be referred to as a voice codec mode supported by the switching target side.

In step 502, the target MSC will obtain the list obtained in step 502 (V, W, X, Y, Z) are sent to the relay MSC corresponding to the relay MGW.

The operations performed in step 502 can be performed after the calling user completes the handover.

In step 503, the relay MSC obtains an intersection with the received list (V, W, X, Y, Ζ) according to the voice codec mode supported by the relay MGW managed by itself, and obtains the relay MGW and the calling side. A list of supported voice codec modes (V, W, X, Z), and the list is sent to the called MSC.

In step 504, the called MSC picks up the intersection with the received list according to the voice codec mode supported by the called side, and selects the voice codec mode with the highest priority. When the received list is (V, W, X, Z), the codec mode supported by the called side is (Y, V, Ζ). After the called MSC takes the intersection, the result of the selection is V. The called MSC sends the selected result to the relay MSC, and notifies the called MGW and the called BSC.

Here, the voice codec mode supported by the called side is: the intersection of the called user terminal, the base station controller or the radio network controller where the called user terminal is located, and the voice codec mode supported by the MGW where the called user terminal is located. . The voice codec list supported by the called user terminal may be adjusted according to the resource optimization principle, or may not be adjusted.

In step 505, the relay MSC sends the received voice codec mode to the target MSC, and notifies the relay MGW.

In step 506, the target MSC receives the voice codec mode sent by the relay MSC. Since the called MSC selects the voice codec mode V at this time, the target MSC does not need to notify the voice codec used by the current user terminal of the target MGW. the way. However, if the voice codec mode supported by the called side is (Y, X, Ζ), the voice codec mode selected by the called MSC is X, since the target MGW and the target BSC are currently using V. Therefore, the target MSC needs to further reconfigure the voice codec mode currently used by the target MGW and the target BSC. Specifically, as shown by the dashed arrow in step 506 of FIG. When the target BSC, the target MGW, the relay MGW, the called BSC, and the called MGW both obtain the voice codec mode used by the current calling user, the target BSC, the target MGW, the relay MGW, the called BSC, and the Calling MGW, you can set up the call of the same voice codec to achieve TrFO.

The calling user terminal mentioned in the above description is a user terminal that has switched during the call, so that the user terminal that has changed the handover can be referred to as a handover user terminal; and during the call, no handover occurs. The user terminal is referred to as a non-handover user terminal, that is, the called user terminal described in the second preferred embodiment. The flow shown in Figure 5 is not limited to the execution when the calling user terminal switches. When the called user terminal switches, the steps shown in Figure 5 can also be performed. At this time, the called user terminal is a handover user terminal, and the calling user terminal is a non-handover user terminal.

In the flow shown in Fig. 5, the flow of the method of the embodiment of the present invention performed by the user at the time of switching is introduced. When the user establishes a call, the voice codec mode used in common can also be negotiated between the master/called party according to the flow shown in FIG. However, during the call setup process, the list of voice codec modes supported by the calling side is supported by the calling user terminal, the base station controller or the radio network controller where the calling user terminal is located, and the MGW where the calling user terminal is located. The voice codec mode; and the MSC that determines the voice codec mode used by the user terminal is the MSC where the calling user terminal is currently located. In the same manner, the embodiment of the present invention can adjust the voice codec mode of the user terminal in the process of call setup, and negotiate and determine the voice codec mode used by the master/called user terminal. Meanwhile, in the process of the handover, the renegotiation determines the master. / The voice codec mode used by the called user terminal. It is also possible to negotiate to determine the voice codec mode used by the master/user terminal only during the handover process.

The above describes the case where there is one relay MGW. When there are multiple relay MGWs, the relay MSC corresponding to the multiple relay MGWs also negotiates the intersection.

When the master/called party is located in the same MGW, the user terminal that has switched needs only needs to cooperate with itself. The MGW in which it is located takes the intersection. When the master/called party is located in the two MGWs, the MSC on the calling side only needs to send the MGW supported by the calling side to the called side MSC, and the called MSC according to the list of voice codec modes supported by the called user terminal side, And the received intersection of the list, determine the voice codec mode used by the calling and called user terminals, and return to the MSC where the calling user terminal is located.

Further, the method of storing the speech codec mode described in the first preferred embodiment is also applicable to the second preferred embodiment.

Referring to FIG. 6, FIG. 6 is a structural diagram of a mobile switching center according to a second preferred embodiment of the present invention. The block diagram includes an adjustment unit 61, an intersection unit 63, an execution unit 62, a list receiving unit 64, and a list transmitting unit 65.

The adjusting unit 61 is mainly configured to adjust the priority of the received voice codec mode supported by the user terminal according to the resource optimal principle. The intersection unit 63 is mainly used to intersect the voice codec mode supported by the base station controller or the radio network controller with the voice codec mode supported by the user terminal adjusted by the adjustment unit 61; and send the list obtained by the intersection to the execution unit. 62. Correspondingly, the executing unit 62 determines the voice codec mode according to the received list.

In addition, in order to implement TrFO, the voice codec mode between the master and the called user terminal is consistent, and further includes: a list receiving unit 64 and a list sending unit 65. The list receiving unit 64 is configured to receive a list of voice codec modes sent by the upstream mobile switching center, and send the received list to the intersection unit 63.

The intersection unit 63 is configured to: cross the received list with a list of voice codec modes supported by the MGW managed by the mobile switching center, and send the obtained list to the downstream mobile switching center by using the list sending unit 65. When the intersection unit 63 sends the list after the intersection to the downstream mobile switching center, the current MSC plays the corresponding role in the current MSC.

The intersection unit 63 can also control the received user list and the base station corresponding to the list. The voice codec mode supported by the controller or the radio network controller, or the MGW takes the intersection, and the result of the intersection is sent to the execution unit 62. At this time, the corresponding role of the current MSC in the current call is the MSC corresponding to the called user terminal or the MSC corresponding to the non-handover party.

In addition, the intersection unit 63 may further transmit the voice codec mode supported by the user terminal adjusted by the adjustment unit 61 to the voice codec mode supported by the base station controller or the radio network controller, and send the message to the downstream mobile switch through the list sending unit 65. center. At this time, the corresponding role of the current MSC in the current call is the MSC corresponding to the user terminal or the MSC corresponding to the handover party.

The list transmitting unit 65 is mainly used to transmit the list according to the indication of the intersection unit 63. The technical solution of the embodiment of the present invention adjusts the priority of the user terminal according to the resource optimization principle, so that the voice codec mode used by the user terminal occupies less network resources than the voice codec mode used in the past. The technical solution of the embodiment of the invention not only saves the bearer network resources, but also saves the air interface resources.

In the technical solution of the embodiment of the present invention, the core network adjusts the priority order of the voice codec mode supported by the user terminal according to the resource optimization principle, so that the voice codec mode resource used by the user terminal is minimized, thereby maximally saving the bearer network. The resources, especially the bandwidth, and the implementation of the embodiment of the present invention do not affect the normal service of the user terminal user. In the meantime, the technical solution of the embodiment of the present invention enables the primary and called user terminals in the call process to use the same voice codec when the A interface is IPized, and implements TrFO. Moreover, when the calling user terminal or the called user terminal is switched, the same codec mode can be ensured for the main and called user terminals, especially after the TrFO 3G network is switched to the 2G network. The TrFO in the main/called call process saves valuable codec (TC) resources, avoids the inter-system handover failure caused by the codec conversion, and provides the success rate of the handover. The MSC corresponding to GSM in the WCDMA system is referred to as an MSC Server. In the present invention, the MSC and the MSC Server are collectively referred to as a mobile switching center; in addition, the WCDMA system corresponds to a base station controller as a radio network controller.

The above are only the preferred embodiments of the present invention, and are not intended to limit the present invention. Any modifications, equivalents, improvements, etc., which are within the spirit and scope of the present invention, should be included in the protection of the present invention. Within the scope.

Claims

Claim

A method for adjusting a voice codec mode, the method comprising the steps of:

After receiving the call initiated by the user terminal, adjusting the priority of the voice codec mode according to the voice codec mode reported by the user terminal, so that the voice codec mode with low resource occupation has a higher priority;

The voice encoding and decoding mode used by the user terminal is determined according to the priority of the adjusted voice codec mode.

2. The method of claim 1 wherein

The method further includes: saving a priority of the adjusted voice codec mode; determining, in the subsequent call of the user terminal, a voice codec mode used by the user terminal according to a priority of the saved voice codec mode.

The method according to claim 1 or 2, wherein the determining, according to the priority of the adjusted voice codec mode, determining a voice codec mode used by the user terminal is: determining that the user terminal uses the highest priority Voice codec mode.

The method according to claim 1 or 2, wherein, according to the voice codec mode reported by the user terminal, adjusting the priority of the voice codec mode to give priority to the voice codec mode with low resource occupation Levels include:

According to the network resources required by the voice codec mode, the voice codec mode that occupies the smallest network resource is adjusted to the highest priority.

The method according to claim 1 or 2, wherein, according to the voice codec mode reported by the user terminal, the priority of the voice codec mode is adjusted to give priority to the voice codec mode with low resource occupation Levels include:

Adjusting the voice supported by the user terminal according to the user priority set for the user terminal The priority of the codec mode is such that the voice codec mode with low resource occupation has a high priority.

The method according to claim 5, wherein the adjusting the priority of the voice codec mode supported by the user terminal according to the user priority set by the user terminal, so that the voice codec mode with low resource occupation is adopted The high priority includes:

For the user terminal with the highest user priority, adjust the voice codec mode with the highest network resource occupancy to the highest priority; for the user terminal without the highest user priority, adjust the voice codec mode with the smallest network resource occupancy. To the highest priority.

The method according to claim 1 or 2, wherein the method further comprises:

And the voice codec mode supported by the base station controller or the radio network controller and the voice codec mode reported by the user terminal are intersected;

Adjusting the priority of the voice codec mode included in the intersection is such that the voice codec mode with low resource occupancy has a high priority.

The method according to claim 1 or 2, wherein the method further comprises:

And the voice codec mode supported by the media gateway and the voice codec mode reported by the user terminal are intersected;

The method according to claim 1, wherein the determining, according to the priority of the adjusted voice codec mode, the voice codec mode used by the user terminal comprises: in the handover process, the target MSC where the user terminal is located The voice codec mode used by the user terminal is determined according to the intersection of the voice codec mode supported by the target base station controller or the radio network controller and the voice codec mode supported by the user terminal.

10. The method of claim 1 wherein: After receiving the call initiated by the user terminal, adjusting the priority of the voice codec mode according to the voice codec mode reported by the user terminal, so that the voice codec mode with a low resource occupation has a high priority includes: After receiving the call of the first user terminal or the first user terminal in the call setup process, adjusting a voice codec mode of the first user terminal and/or the second user terminal;

Determining, according to the priority of the adjusted voice codec mode, the voice codec mode used by the user terminal, including: a list of voice codec modes supported by the first user terminal side of the first user terminal, and the second user terminal The intersection of the list of voice codec modes supported by the second user terminal side determines the voice codec mode used by the first and second user terminals.

The method according to claim 10, wherein the list of voice codec modes supported by the first user terminal side of the first user terminal and the second user terminal side of the second user terminal are located The intersection of the supported voice codec modes, determining the voice codec modes used by the first and second user terminals includes:

The MSC where the first user terminal is located sends a list of voice codec modes supported by the first user terminal side to the MSC where the second user terminal is located;

The MSC where the second user terminal is located determines the voice codec mode used by the first and second user terminals according to the list of voice codec modes supported by the second user terminal side and the received intersection of the list, and returns to the first The MSC where the user terminal is located.

The method according to claim 10, wherein, when there is a relay MGW, the list of voice codec modes supported by the first user terminal side where the first user terminal is located, and the second user terminal The voice codec mode of the first and second user terminals is determined by the intersection of the list of voice codec modes supported by the second user terminal. The MSC where the first user terminal is located is the relay MSC where the relay MGW is located. Sending a list of voice codec modes supported by the calling side; The relay MSC crosses the received list with the voice codec mode supported by the relay MGW, and obtains a list of voice codec modes supported by the relay MGW and the first user terminal side; The list is sent to the MSC where the second user terminal is located; the MSC where the second user terminal is located determines the first according to the list of voice codec modes supported by the second user terminal side and the list received from the relay MSC. And the voice codec mode used by the two user terminals is returned to the MSC where the first user terminal is located.

The method according to claim 10, 11 or 12, wherein, in the handover process, performing the priority according to the adjusted voice codec mode, and determining a voice codec mode used by the user terminal;

The first user terminal is a handover user terminal that has a handover; the second user terminal is a non-user handover user terminal that does not have a handover; the list of voice codec modes supported by the first user terminal side is a handover user. a terminal, a target base station controller or a radio network controller to which the switching user terminal is switched, and a voice codec mode supported by the target MGW to which the switching user terminal is switched; the list of voice codec modes supported by the non-switching side is The voice codec mode supported by the non-switching user terminal, the base station controller or the radio network controller where the non-switching user terminal is located, and the MGW where the non-switching user terminal is located.

A device for adjusting a voice codec mode, the device comprising: an adjusting unit, configured to adjust a priority of the voice codec mode to make a voice codec mode with a low resource occupation have a high priority;

15. Apparatus according to claim 14 wherein:

The adjusting unit is configured to adjust a priority of a voice codec mode supported by the user terminal according to a user priority of the user terminal.

16. Apparatus according to claim 14 wherein: The adjusting unit is configured to perform an intersection of a voice codec mode supported by the user terminal according to a voice codec mode supported by the base station controller, and/or a voice supported by the user terminal according to a voice codec mode supported by the media gateway. The codec method takes the intersection and obtains the voice codec mode finally supported by the user terminal.

The device according to claim 14, 15 or 16, wherein the device further comprises: a control unit and a storage unit;

The storage unit is located between the adjustment unit and the execution unit, and receives and saves the result of adjusting the voice codec mode of the user terminal by the adjustment unit;

The control unit is configured to receive a voice encoding and decoding method supported by the user terminal, and determine whether the corresponding adjustment result is saved in the storage unit. When the storage unit saves the corresponding adjustment result, the instruction execution unit is configured according to the storage unit. Adjusting the result, performing the operation of determining the voice codec mode used by the user terminal; and when the storage unit does not save the corresponding adjustment result, instructing the adjustment unit to perform the operation of the adjustment.

18. A mobile switching center, the mobile switching center comprising: at least: an adjusting unit, an executing unit, and an intersection unit;

The adjusting unit is configured to adjust a priority of a voice codec mode supported by the user terminal, so that a voice codec mode with a low resource occupation has a high priority;

The intersection unit is configured to: cross the voice codec mode supported by the base station controller or the radio network controller, and the voice codec mode supported by the user terminal adjusted by the adjustment unit; and obtain the list obtained by the intersection Sent to the execution unit;

The mobile switching center according to claim 18, wherein the mobile switching center further comprises: a list receiving unit and a list transmitting unit;

The list receiving unit is configured to receive a voice codec sent by an upstream mobile switching center a list of modes, and send the received list to the intersection unit;

The intersection unit is configured to: cross the received list with a list of voice codec modes supported by the MGW managed by the mobile switching center, and send the obtained list to the downstream mobile exchange by using the list sending unit. Centering; or, the received list is intersected with a user terminal, a base station controller, or a radio network controller corresponding to the list, or a voice codec mode supported by the MGW, and the result of the intersection is sent to the execution unit. Or, the intersection of the voice codec mode supported by the user terminal adjusted by the adjustment unit and the voice codec mode supported by the base station controller or the radio network controller is sent to the downstream mobile switching center through the list sending unit;

The list sending unit is configured to send the list according to the indication of the intersection unit.