CN1747465A - Realization of speech service - Google Patents

Abstract

The invention discloses a method for implementing voice services. The method is to set a VDP layer on the RAN for mutual conversion between voice frame data and voice packet data, and the sender RAN receives the coded voice from the sender MS. After frame data, the voice frame data is converted into voice packet data by its own VDP layer and sent to the packet network; after receiving the voice packet data from the packet network, the receiver RAN converts the voice packet data into voice packet data by its own VDP layer voice frame data, and send the voice frame data to the receiver MS, and the receiver MS decodes the voice frame data and plays it. After applying the method of the present invention, the RAN is used to convert between voice frame data and voice packet data, without setting a special packet gateway device for switching between voice frames and voice packets in the system, or adding voice frames on the terminal And voice packet conversion function, thereby reducing the cost of the system and simplifying the structure of the system.

Description

A method for realizing voice service

technical field

The invention relates to voice transmission technology, in particular to a method for realizing voice services in a packet network.

Background technique

Traditional circuit-based voice transmission requires a bandwidth of 64kbps. With the development of 2.5G and 3G wireless networks, packet data services are introduced. Packet voice technology usually adopts a coding method less than 8Kbps. The actual average bandwidth of a network for packet voice tends to be as low as 2 to 3Kbps. Therefore, the packet voice technology can effectively save bandwidth, thereby reducing the networking and operating costs of the packet network, and, because the packet network has the characteristics of flexible networking, it has attracted more and more attention, so it can be widely used in wireless networks .

In the system for implementing packet voice services, it is necessary to set up a unit for converting voice frame data and voice packet data to perform conversion between voice frame data and voice packet data. In addition, there is packet jitter in the packet network, and the size of the jitter is reflected within a certain period of time. In terms of the difference in end-to-end delay of different packet data, the greater the difference in end-to-end delay between packet data, the packet The more serious the jitter of the network is. When using the packet network to transmit voice data, the jitter will cause the voice data to not be played in time, causing sound vacancy, thereby affecting the voice quality of the call. Therefore, it is necessary to perform jitter processing on the voice packet data.

There are many ways to realize packet voice service in CDMA system, among them, the system structure of a more traditional one to realize packet voice service is shown in Figure 1. Encoded on the mobile station (MS) and sent from the sender MS to the sender packet voice gateway via the radio access network (RAN) and mobile switching center (MSC) in the format of voice frame data, and the voice frame is transmitted by the sender packet voice gateway The data is converted into voice packet data, which is transmitted from the packet network to the receiver. The receiver’s packet voice gateway performs jitter processing on the voice packet data from the packet network and converts the voice packet data into voice frame data, and then sends it to the receiver through MSC and RAN. The party MS and the receiver MS decode the voice frame data into voice signals.

The disadvantage of this networking method is that it is necessary to set up a special packet voice gateway device to convert voice frame data and voice packet data, and perform jitter processing, which increases additional costs.

In order to solve the defects of the above packet voice implementation method, another CDMA system has emerged to realize the packet voice service. The system structure adopted by this packet voice service is shown in Figure 2. The packet voice service in this networking mode The specific implementation method is: directly pack the voice data into voice packet data on the sender MS, send it to the packet network through the sender RAN, the packet network transmits the voice packet data to the receiver RAN, and the receiver RAN sends the voice packet data to the receiver The party MS and the receiver MS perform jitter processing on the voice packet data and directly decode the data into a voice signal.

It can be seen that compared with the first networking method, the second networking method does not need to set up a special packet voice gateway. However, this networking method also has some defects:

First of all, the conversion between voice packet data and voice frame data needs to be realized by the mobile phone, and most early mobile phones cannot support this function.

Secondly, this networking method requires the mobile phone to perform jitter processing, which leads to complex processing of the mobile phone, requires higher processing performance, and increases the cost of the mobile phone.

Contents of the invention

The main purpose of the present invention is to provide a method for implementing voice services, and the RAN realizes the conversion between voice frame data and voice packet data.

The purpose of the present invention is achieved through the following technical solutions:

A method for realizing a voice service, the method comprising the steps of:

Add a voice data protocol VDP layer for mutual conversion between voice packet data and voice frame data on the protocol layer of the radio access network RAN;

After the sender RAN receives the encoded voice frame data from the sender mobile station MS, its own VDP layer converts the voice frame data into voice packet data, and sends the voice packet data to the packet network;

After receiving the voice packet data from the packet network, the receiver RAN converts the voice packet data into voice frame data by its own VDP layer, and sends the voice frame data to the receiver MS, and the receiver MS decodes and plays the voice frame data .

Wherein, the method further includes: allocating a jitter buffer with a certain capacity for temporarily storing voice packet data in the RAN for each connected link,

After the receiving side RAN receives the voice packet data, before converting the voice packet data into voice frame data, it further includes: the receiving side RAN stores the received voice packet data into the jitter buffer, and when the fetching condition is satisfied, from The jitter buffer takes voice packet data to the VDP layer.

Wherein, the method for the receiver RAN to store the received voice packet data into the jitter buffer is: the receiver RAN judges whether the current jitter buffer has remaining space, and if so, stores the voice packet data into the jitter buffer; Otherwise, discard the earliest voice packet data stored in the jitter buffer, and then store the received voice packet data into the jitter buffer.

Wherein, the method for the receiver RAN to take out the voice packet data from the jitter buffer to the VDP layer is:

A, the receiver RAN judges whether the jitter buffer is empty, if yes, then proceeds to step B; otherwise, takes out the voice packet data from the jitter buffer to the VDP layer, then returns to step A, and continues to judge whether the jitter buffer is empty;

B. The receiver RAN judges whether the condition for taking out data from the jitter buffer is satisfied. If yes, it takes out the voice packet data from the jitter buffer to the VDP layer, and then returns to step A; otherwise, returns to step B and continues to judge whether the current condition is satisfied. Conditions for fetching data from the jitter buffer.

Wherein, the method for judging whether the condition for fetching data from the jitter buffer is currently met is: the receiver RAN judges whether the space occupancy rate of the jitter buffer reaches the set buffer space occupancy threshold, and if so, then from the jitter buffer area to take out the voice packet data to the VDP layer; otherwise, continue to judge whether the space occupancy rate of the jitter buffer reaches the set buffer space occupancy rate threshold.

Wherein, the method further includes: setting a jitter buffer timer in the RAN,

In step B, before judging whether the data is taken out from the jitter buffer, it further includes: starting the jitter buffer timer,

The method for judging whether the condition for taking out data from the jitter buffer is satisfied at present is as follows:

B01, the receiver RAN judges whether the space occupancy rate of the jitter buffer reaches the set buffer space occupancy threshold, if yes, then takes out the voice packet data from the jitter buffer to the VDP layer, and then exits this process; otherwise, go to step B02;

B02. The receiver RAN judges whether the jitter buffer timer has expired, and if so, takes out the voice packet data from the jitter buffer to the VDP layer, and then exits this process; otherwise, returns to step B01.

Wherein, the method further includes: setting a jitter buffer timer in the RAN,

In step B, before judging whether the data is taken out from the jitter buffer, it further includes: starting the jitter buffer timer,

The method for judging whether the condition for taking out data from the jitter buffer is satisfied at present is as follows:

B11, the receiving side RAN judges whether the jitter buffer timer is overtime, if yes, then take out the voice packet data from the jitter buffer to the VDP layer, then exit this process; otherwise, go to step B12;

B12, the receiver RAN judges whether the space occupancy rate of the jitter buffer reaches the set buffer space occupancy threshold, if yes, then takes out the voice packet data from the jitter buffer to the VDP layer, and then exits this process; otherwise, returns to step B11 .

Wherein, the capacity of the jitter buffer is a fixed value set according to the predicted jitter of the packet network and the call quality required by the user.

Wherein, the capacity of the jitter buffer is dynamically determined by the receiving RAN according to the current jitter situation of the packet network.

Wherein, the method for dynamically determining the capacity of the jitter buffer according to the jitter situation of the current packet network is:

Determine the jitter value of the current voice packet data reflecting the jitter situation of the current packet network;

Determine the capacity of the current jitter buffer according to the jitter value of the current voice packet data.

Wherein, the method for determining the jitter value of the current voice packet data is: judging whether the current voice packet data is the first voice packet data received by the receiver RAN in this voice service, and if so, the receiver RAN will have The set initial jitter value is used as the jitter value of the current voice packet data; otherwise, the receiver RAN determines the transmission time of the current voice packet data, and then according to the transmission time of the current voice packet data, the transmission time of the last voice packet data and the last voice The jitter value of the packet determines the jitter value of the current packet.

Wherein, the method for determining the transmission time of the voice packet data is: the receiver RAN subtracts the data generation time contained in the data from the time when the voice packet data is received by the receiving RAN to obtain the transmission time of the voice packet data.

Wherein, the method for determining the jitter value of the current voice packet data according to the transmission time of the current voice packet data, the transmission time of the last voice packet data and the jitter value of the last voice packet data is:

Jitter value _n = Jitter value _n-1 + (|transmission time _n - transmission time _n-1 |-jitter value _n-1) /o',

Among them, the jitter value _n is the jitter value of the current voice packet data, the jitter value _n-1 is the jitter value of the last voice packet data, the transmission time _{n is} the transmission time of the current voice packet data, and the transmission time _n-1 is the last voice The transmission time of packet data, o' is the convergence coefficient.

Wherein, the convergence coefficient is 16.

Wherein, the method for determining the jitter buffer capacity according to the jitter value of the current voice packet data is:

Set the maximum jitter threshold, minimum jitter threshold, maximum capacity threshold and minimum capacity threshold on the RAN;

Compare the jitter value of the current voice packet data with the maximum jitter threshold and the minimum jitter threshold, if the jitter value of the current voice packet data is greater than the maximum jitter threshold, then judge whether the current jitter buffer capacity is less than the maximum capacity threshold, if yes, then Increase the jitter buffer capacity; otherwise, keep the current jitter buffer capacity unchanged;

If the jitter value of current voice packet data is less than the minimum jitter threshold, then judge whether the current jitter buffer capacity is greater than the minimum capacity threshold, if so, then reduce the jitter buffer capacity; otherwise, keep the current jitter buffer capacity unchanged;

If the jitter value of the current voice packet data is greater than or equal to the minimum jitter threshold and less than or equal to the maximum jitter threshold, keep the capacity of the current jitter buffer unchanged.

Wherein, the method for increasing the capacity of the jitter buffer is:

Set the capacity step value on the RAN;

The increased jitter buffer capacity is the sum of the current jitter buffer capacity and the capacity step value,

The method for reducing the capacity of the jitter buffer is as follows:

Set the capacity step value on the RAN;

The reduced jitter buffer capacity is the difference between the current jitter buffer capacity and the capacity step value.

Wherein, the method for increasing the capacity of the jitter buffer is:

Set capacity percentage on RAN;

The increased jitter buffer capacity is the current jitter buffer capacity plus the product of this capacity and the capacity percentage,

The method for reducing the capacity of the jitter buffer is as follows:

Set capacity percentage on RAN;

The reduced jitter buffer capacity is the current jitter buffer capacity minus the product of this capacity and the capacity percentage.

Wherein, the method further includes: allocating a jitter buffer of a certain capacity in the RAN to temporarily store voice frame data,

After the receiver RAN converts the voice packet data into voice frame data, it further includes:

The receiver RAN stores the voice frame data in the jitter buffer;

The receiving RAN fetches voice frame data from the jitter buffer.

Wherein, the packet network is a code division multiple access network.

The present invention provides a method for realizing voice service, the method is to add a VDP layer for mutual conversion between voice packet data and voice frame data on the protocol layer of RAN; After encoding the voice frame data, the voice frame data is converted into voice packet data by its own VDP layer, and the voice packet data is sent to the packet network; when the receiver RAN receives the voice packet data from the packet network, The layer converts the voice packet data into voice frame data, and sends the voice frame data to the receiver MS, and the receiver MS decodes and plays the voice frame data. In the prior art method, the packet voice gateway realizes the mutual conversion between voice packet data and voice frame data, or the mobile phone realizes the conversion between voice packet data and voice frame data. From the comparison between the present invention and the prior art, the method of the present invention does not need to set special equipment to convert between voice packet data and voice frame data, thereby simplifying the network structure and avoiding additional costs; in addition, the present invention does not require The mobile phone performs the conversion of the above data format, so that there is no additional requirement on the mobile phone, and the operation of the mobile phone is simplified at the same time.

Moreover, it can be seen from the technical solution that in the method of the present invention, the receiver RAN performs jitter processing on the voice packet data, which also avoids the increase of network cost or the increase in network cost caused by the jitter processing of the packet voice gateway or mobile phone in the prior art. Complicated issues with mobile phone operations.

Description of drawings

FIG. 1 is a schematic structural diagram of a system for implementing voice services in prior art 1;

FIG. 2 is a schematic structural diagram of a system for implementing voice services in the second prior art;

FIG. 3 is a schematic structural diagram of the protocol layer of the RAN of the present invention;

Fig. 4 is a schematic structural diagram of a system for realizing a voice service according to the present invention;

Fig. 5 is the flow chart of the method for realizing voice service according to the present invention;

Fig. 6 is a flowchart of storing voice packet data into the jitter buffer by the receiving RAN.

Fig. 7 is a flow chart of the receiving RAN fetching the voice packet data from the jitter buffer.

Fig. 8 is a flowchart of a method for dynamically adjusting the size of a jitter buffer according to the present invention.

Detailed ways

In order to make the purpose, technical solution and advantages of the present invention clearer, the present invention will be further described below in conjunction with the accompanying drawings and specific embodiments.

The method of the present invention improves the prior art on the basis of the networking mode shown in Figure 2, and increases the voice data protocol (VDP) layer on the RAN, and Figure 3 is a structural representation of the protocol layer of the RAN of the present invention, from It can be seen from Fig. 3 that in the protocol layer interacting with the packet network, the RAN of the present invention, in addition to including the original packet transport layer, link layer and physical layer, also adds a VDP layer above the packet transport layer , to realize the conversion between voice frame data and voice packet data. The RAN has the function of converting speech frame data into speech packet data and converting speech packet data into speech frame data.

Fig. 4 is a schematic structural diagram of a system for realizing voice services according to the present invention. The method of the present invention transmits voice data to the sender RAN in the format of voice frames on the sender MS, and the sender RAN converts the voice frame format into a voice packet format , and then sent to the packet network, the packet network transmits the voice packet data to the receiver RAN, and the receiver RAN performs jitter processing on the received voice packet data, and then converts the voice packet data into voice frame data, and sends it to the receiver MS , the receiver MS directly decodes the data in the voice frame format into a voice signal.

The jitter processing is that the receiver RAN stores the received voice packet data into the preset jitter buffer, takes out the voice packet data from the jitter buffer when certain extraction conditions are met, and then executes the process of converting the voice packet data into voice frame data step.

Fig. 5 is the flow chart of the method for realizing voice service according to the present invention, as can be seen from Fig. 5, the method of the present invention comprises the following steps:

Step 501: Add a VDP layer and a jitter buffer for jitter processing on the sending RAN and the receiving RAN respectively, and set the space occupancy threshold and buffer timer of the jitter buffer.

The role of the VDP layer is to realize the mutual conversion between voice frame data and packet data. Adding the VDP layer on the RAN can make the RAN have the function of converting voice frame data into voice packet data and converting voice packet data into voice frame data. .

Step 502: The sender MS encodes the voice data of the user, converts the voice into a digital bit stream, and sends the voice data to the RAN in the form of voice frames over the air interface.

There are many ways to encode voice data, for example, two types of encoding can be adopted: EVRC (Enhanced Variable Rate Codec) or QCELP (QUALCOMM Code Excited Linear Prediction) proposed by Qualcomm Way. The speech frame data can also adopt different formats, for example, the format of 20ms/frame can be adopted, that is, the speech data of 20ms is stored in each frame.

Step 503: After receiving the voice frame data sent by the sender MS, the sender RAN converts the voice frame data into voice packet data according to a certain voice packet method at the VDP layer, fills in the destination address, and then transmits the voice packet data packet network.

There are many voice grouping methods. For example, one voice frame can be packaged into one voice group, or multiple voice frames can be packaged into one voice group. The number of voice frames contained in each voice group is determined by the set parameters, and The number has its own advantages and disadvantages: the fewer voice frames each voice packet contains, the smaller the network delay, but the larger the number of voice packets, the larger the burden on the packet network; the more voice frames each voice packet contains, the slower the network delay will be. The longer the delay, the smaller the number of voice packets, so the burden on the packet network is smaller. Due to the above advantages and disadvantages, it is necessary to dynamically adjust the parameters controlling the voice grouping mode according to the voice quality and network load measured on the network at different times, so as to determine the number of voice frames for forming a voice group.

Step 504: The receiver RAN uses the set jitter buffer to perform jitter processing on the voice packet data sent by the packet network.

Step 505: the receiver RAN converts the voice packet data into voice frame data at the VDP layer, and sends the voice frame data to the receiver MS.

Step 506: The receiver MS decodes the received voice frame data according to the frame format of the sender MS, and plays the voice data.

When the receiving RAN does not send voice data to the MS, in order not to make the user of the receiving MS feel obvious voice interruption, it can send comfort noise to the receiving MS, and the receiving MS decodes the comfort noise and plays the sound. The comfort noise is Refers to the special noise that does not stimulate the human body.

In step 504, the receiving side RAN performs jitter processing on the voice packet data sent by the packet network, and the flow process of the jitter processing is composed of two parallel processes, one is that the receiving side RAN stores the voice packet data into the jitter buffer, The other is the process that the receiver RAN takes out voice packet data from the jitter buffer. The two processes of dithering processing are introduced in detail below.

Fig. 6 is the flow chart of receiver RAN storing voice packet data into the jitter buffer, as can be seen from the figure, the process includes the following steps:

Step 601: The receiver RAN judges whether it has received voice packet data from the packet network, and if yes, then proceeds to step 602; otherwise, returns to step 601, and continues to judge whether voice packet data is received.

Step 602: the receiving RAN judges whether there is remaining space in the jitter buffer, if yes, go to step 604; otherwise, go to step 603.

Step 603: the receiver RAN discards the earliest voice packet data stored in the jitter buffer, and then goes to step 604.

Step 604: the receiver RAN stores the voice packet data in the jitter buffer, and then repeatedly executes step 601.

Fig. 7 is the flow chart of receiving side RAN taking out voice packet data from the jitter buffer, as can be seen from the figure, the process includes the following steps:

Step 701: The receiving RAN judges whether the jitter buffer is empty, if yes, go to step 702; otherwise, go to step 704.

Step 702: The receiving RAN starts the jitter buffer timer.

Step 703: The receiver RAN judges whether the condition for taking out the voice packet data from the jitter buffer is satisfied at present. There are two judgment methods, one of which can be selected:

(1) The receiver RAN judges whether the space occupancy rate of the jitter buffer has reached the space occupancy threshold of the buffer capacity set, if it reaches the threshold, then go to step 704; if it does not reach the threshold, then judge the jitter buffer timer Whether it is overtime, if overtime, go to step 704; if not overtime, then return to step 703.

(2) the receiver RAN judges whether the jitter buffer timer is overtime, if overtime, then go to step 704; if not overtime, then judge whether the space occupancy rate of the buffer zone reaches the space occupancy rate threshold, if it reaches the threshold value, then go to step 704 704 ; if the threshold is not reached, return to step 703 .

Step 704: The receiver RAN fetches the voice packet data from the jitter buffer to the VDP layer.

It should be noted that in the scheme of extracting voice packet data from the jitter buffer, it is also possible not to set the jitter buffer timer, but directly judge whether the space occupancy rate of the current jitter buffer reaches the space occupancy threshold to determine whether the current condition for extracting voice is satisfied. Criteria for grouping data.

In addition, the capacity of the jitter buffer can be a parameter configured by the network, or a parameter that can be changed dynamically according to the jitter situation of the packet network. The foregoing embodiments belong to the former case. When the network is configured, the capacity of the buffer zone is adjusted according to the predicted jitter situation of the packet network and the voice quality required by the user. Once the configuration is completed, this parameter cannot be automatically adjusted; In one case, it is necessary to set an initial jitter buffer capacity, and then dynamically change the capacity according to the jitter of the packet network. The method of dynamically changing the jitter buffer capacity is: after the receiver RAN receives new voice packet data , calculate the jitter value according to the received voice packet data, and adjust the capacity of the jitter buffer according to the jitter value.

The method for dynamically adjusting the capacity of the jitter buffer is described below.

In order to dynamically adjust the jitter buffer capacity, in addition to adding the VDP layer and jitter buffer on the RAN and setting various parameters, it is also necessary to set some parameters required for dynamically adjusting the jitter buffer capacity, such as initial jitter value, maximum jitter threshold, The minimum jitter threshold, the maximum capacity threshold and the minimum capacity threshold of the jitter buffer.

Since the premise of implementing dynamic adjustment of the jitter buffer is to record the time when the sender RAN generates the voice packet data and the time when the receiver RAN receives the voice packet data, when the sender RAN forms the voice packet data, record the current system time and Add this time as the packet data generation time to the voice packet data; when the receiver RAN receives the voice data from the packet network, record the current system time as the packet data arrival time.

After the receiver RAN receives the voice packet data from the packet network, before performing jitter processing on the voice packet data, the step of dynamically adjusting the buffer capacity is added, as shown in Figure 8, the method of dynamically adjusting the jitter buffer capacity of the present invention Proceed as follows:

Step 801: The receiver RAN calculates the jitter value according to formula (1):

Jitter value _n = Jitter value _n-1 + (|transmission time _n - transmission time _n-1 |-jitter value _n-1 )/o'(1)

Among them, the jitter value _n is the jitter value calculated currently; the jitter value _n-1 is the jitter value calculated last time; the transmission time _n is the transmission time of the current packet data on the network, and its value is the difference between the arrival time of the current packet data and The difference between the generation time of the packet data carried in the packet; the transmission time _n-1 is the transmission time of the last packet data on the network; o' is a convergence coefficient that can be adjusted appropriately according to the network situation, preferably, its value can be 16.

If the currently received voice packet data is the first voice packet data, first calculate the transmission time of the packet data. The transmission time is the difference between the arrival time of the packet data and the generation time of the packet data. At this time, the jitter value is not calculated, and the jitter value is set initial jitter value.

After receiving the second voice packet data from the receiver RAN, the transmission time of the last voice packet data is taken as the transmission time _n-1 , the jitter value of the last voice packet data is taken as the jitter value _n-1 , and the current packet The transmission time of the data is taken as the transmission time _n , and the jitter value _n is calculated with the formula (1).

Step 802: Compare the calculated jitter value at the current moment with the maximum jitter threshold and the minimum jitter threshold, if the jitter value is greater than the maximum jitter threshold, go to step 803; if the jitter value is smaller than the minimum jitter threshold, then go to step 804 ; If the jitter value is between the maximum jitter threshold and the minimum jitter threshold, including the maximum jitter threshold and the minimum jitter threshold, go to step 805 ;

Step 803: Judging whether the capacity of the current jitter buffer is smaller than the maximum capacity threshold, if yes, go to step 806; otherwise, go to step 805.

Step 804: Judging whether the capacity of the current jitter buffer is greater than the minimum capacity threshold, if yes, go to step 807; otherwise, go to step 805.

Step 805: Keep the current capacity of the jitter buffer unchanged, and then end.

Step 806: Increase the capacity of the buffer to be jittered, and then end.

There are two ways to increase capacity: one is to increase a fixed step value; the other is to increase a certain percentage of the current capacity, for example, increase 20% of the current capacity.

Step 807: Reduce the capacity of the buffer to be jittered.

There are two ways to reduce the capacity: one is to reduce the fixed step value; the other is to reduce a certain percentage of the current capacity, for example, reduce the current capacity by 20%.

After determining the buffer capacity, the receiver RAN uses the jitter buffer to perform jitter processing on the voice packet data at the VDP layer.

In the above-mentioned method for implementing packet voice services, the receiving RAN performs dithering processing on the received voice packet data, and then converts the voice packet data into voice frame data. In actual processing, the receiver RAN can also first convert the received voice packet data into voice frame data, and then perform dithering processing on the voice frame data. The method of dithering voice frame data is also to use the dithering buffer for dithering processing , in this case, voice frame data is stored in the jitter buffer instead of voice packet data. The capacity of the jitter buffer can be a preset fixed value, or as mentioned above, the jitter value can be calculated according to the current jitter situation of the packet network, and the capacity of the jitter buffer can be dynamically changed according to the calculated jitter value.

Appropriate improvements can be made to the method according to the present invention in the specific implementation process to meet the specific needs of specific situations. Therefore, it can be understood that the specific implementation mode according to the present invention is only for demonstration, and is not intended to limit the scope of protection of the present invention. For example, the present invention is not limited to application in CDMA networks, and can also be applied in other types of packet networks.

Claims (19)

1. A method for realizing voice service, characterized in that the method comprises the steps: Add a voice data protocol VDP layer for mutual conversion between voice packet data and voice frame data on the protocol layer of the radio access network RAN; After the sender RAN receives the encoded voice frame data from the sender mobile station MS, its own VDP layer converts the voice frame data into voice packet data, and sends the voice packet data to the packet network; After receiving the voice packet data from the packet network, the receiver RAN converts the voice packet data into voice frame data by its own VDP layer, and sends the voice frame data to the receiver MS, and the receiver MS decodes and plays the voice frame data . 2. The method for realizing voice services according to claim 1, characterized in that the method further comprises: in the RAN, allocate a jitter buffer with a certain capacity for temporarily storing voice packet data for each connected link, After the receiving side RAN receives the voice packet data, before converting the voice packet data into voice frame data, it further includes: the receiving side RAN stores the received voice packet data into the jitter buffer, and when the fetching condition is satisfied, from The jitter buffer takes voice packet data to the VDP layer. 3. The method for realizing voice services according to claim 2, characterized in that, the receiver RAN stores the received voice packet data into the jitter buffer as follows: the receiver RAN judges whether the current jitter buffer has The remaining space, if yes, then store the voice packet data into the jitter buffer; otherwise, discard the earliest voice packet data stored in the jitter buffer, and then store the received voice packet data into the jitter buffer. 4. The method for realizing a voice service according to claim 2, wherein the method for the receiver RAN to take out the voice packet data from the jitter buffer to the VDP layer is: A, the receiver RAN judges whether the jitter buffer is empty, if yes, then proceeds to step B; otherwise, takes out the voice packet data from the jitter buffer to the VDP layer, then returns to step A, and continues to judge whether the jitter buffer is empty; B. The receiver RAN judges whether the condition for taking out data from the jitter buffer is satisfied. If yes, it takes out the voice packet data from the jitter buffer to the VDP layer, and then returns to step A; otherwise, returns to step B and continues to judge whether the current condition is satisfied. Conditions for fetching data from the jitter buffer. 5. The method for realizing a voice service according to claim 4, wherein the method for judging whether the condition for taking out data from the jitter buffer is satisfied at present is: the receiver RAN judges whether the space occupancy rate of the jitter buffer reaches The buffer space occupancy rate threshold that has been set, if yes, then take out the voice packet data from the jitter buffer to the VDP layer; otherwise, continue to judge whether the space occupancy rate of the jitter buffer reaches the set buffer space occupancy rate threshold. 6. The method for implementing voice services according to claim 4, characterized in that the method further comprises: setting a jitter buffer timer in the RAN, In step B, before judging whether the data is taken out from the jitter buffer, it further includes: starting the jitter buffer timer, The method for judging whether the condition for taking out data from the jitter buffer is satisfied at present is as follows: B01, the receiver RAN judges whether the space occupancy rate of the jitter buffer reaches the set buffer space occupancy threshold, if yes, then takes out the voice packet data from the jitter buffer to the VDP layer, and then exits this process; otherwise, go to step B02; B02. The receiver RAN judges whether the jitter buffer timer has expired, and if so, takes out the voice packet data from the jitter buffer to the VDP layer, and then exits this process; otherwise, returns to step B01. 7. The method for implementing voice services according to claim 4, characterized in that the method further comprises: setting a jitter buffer timer in the RAN, In step B, before judging whether the data is taken out from the jitter buffer, it further includes: starting the jitter buffer timer, The method for judging whether the condition for taking out data from the jitter buffer is satisfied at present is as follows: B11, the receiving side RAN judges whether the jitter buffer timer is overtime, if yes, then take out the voice packet data from the jitter buffer to the VDP layer, then exit this process; otherwise, go to step B12; B12, the receiver RAN judges whether the space occupancy rate of the jitter buffer reaches the set buffer space occupancy threshold, if yes, then takes out the voice packet data from the jitter buffer to the VDP layer, and then exits this process; otherwise, returns to step B11 . 8. The method for implementing voice services according to claim 2, wherein the capacity of the jitter buffer is a fixed value set according to the predicted jitter of the packet network and the call quality required by the user. 9. The method for implementing voice services according to claim 2, characterized in that the capacity of the jitter buffer is dynamically determined by the receiver RAN according to the current jitter of the packet network. 10. The method for realizing voice service according to claim 9, characterized in that, the method for dynamically determining the capacity of the jitter buffer according to the jitter situation of the current packet network is: Determine the jitter value of the current voice packet data reflecting the jitter situation of the current packet network; Determine the capacity of the current jitter buffer according to the jitter value of the current voice packet data. 11. The method for implementing voice services according to claim 10, characterized in that the method for determining the jitter value of the current voice packet data is: judging whether the current voice packet data is received by the receiver RAN in this voice service If it is, the receiver RAN will use the set initial jitter value as the jitter value of the current voice packet data; otherwise, the receiver RAN determines the transmission time of the current voice packet data, and then according to the current voice packet The transmission time of the data, the transmission time of the last voice packet data and the jitter value of the last voice packet data determine the jitter value of the current packet data. 12. The method for implementing voice services according to claim 11, characterized in that the method for determining the transmission time of the voice packet data is: the receiver RAN subtracts the time of receiving the voice packet data from the time included in the data The data generation time of the data is obtained to obtain the transmission time of the voice packet data. 13. The method for implementing voice services according to claim 11, characterized in that the current voice is determined according to the transmission time of the current voice packet data, the transmission time of the last voice packet data and the jitter value of the last voice packet data The method for grouping data jitter values is: Jitter value _n = Jitter value _n-1 + (|transmission time _n - transmission time _n-1 |-jitter value _n-1 )/σ, Among them, the jitter value _n is the jitter value of the current voice packet data, the jitter value _n-1 is the jitter value of the last voice packet data, the transmission time _{n is} the transmission time of the current voice packet data, and the transmission time _n-1 is the last voice The transmission time of packet data, σ is the convergence coefficient. 14. The method for implementing voice services according to claim 13, wherein the convergence coefficient is 16. 15. The method for implementing voice services according to claim 10, characterized in that the method for determining the capacity of the jitter buffer according to the jitter value of the current voice packet data is: Set the maximum jitter threshold, minimum jitter threshold, maximum capacity threshold and minimum capacity threshold on the RAN; Compare the jitter value of the current voice packet data with the maximum jitter threshold and the minimum jitter threshold, if the jitter value of the current voice packet data is greater than the maximum jitter threshold, then judge whether the current jitter buffer capacity is less than the maximum capacity threshold, if yes, then Increase the jitter buffer capacity; otherwise, keep the current jitter buffer capacity unchanged; If the jitter value of current voice packet data is less than the minimum jitter threshold, then judge whether the current jitter buffer capacity is greater than the minimum capacity threshold, if so, then reduce the jitter buffer capacity; otherwise, keep the current jitter buffer capacity unchanged; If the jitter value of the current voice packet data is greater than or equal to the minimum jitter threshold and less than or equal to the maximum jitter threshold, keep the capacity of the current jitter buffer unchanged. 16. The method for implementing voice services according to claim 15, wherein the method for increasing the capacity of the jitter buffer is: Set the capacity step value on the RAN; The increased jitter buffer capacity is the sum of the current jitter buffer capacity and the capacity step value, The method for reducing the capacity of the jitter buffer is as follows: Set the capacity step value on the RAN; The reduced jitter buffer capacity is the difference between the current jitter buffer capacity and the capacity step value. 17. The method for implementing voice services according to claim 15, wherein the method for increasing the capacity of the jitter buffer is: Set capacity percentage on RAN; The increased jitter buffer capacity is the current jitter buffer capacity plus the product of this capacity and the capacity percentage, The method for reducing the capacity of the jitter buffer is as follows: Set capacity percentage on RAN; The reduced jitter buffer capacity is the current jitter buffer capacity minus the product of this capacity and the capacity percentage. 18. The method for implementing voice services according to claim 1, characterized in that the method further comprises: allocating a jitter buffer with a certain capacity in the RAN to temporarily store voice frame data, After the receiver RAN converts the voice packet data into voice frame data, it further includes: The receiver RAN stores the voice frame data in the jitter buffer; The receiver RAN fetches voice frame data from the jitter buffer. 19. The method for implementing voice services according to claim 1, wherein the packet network is a code division multiple access network.

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