WO2016202147A1 - 一种网络电话voip资源处理方法、装置及设备 - Google Patents

一种网络电话voip资源处理方法、装置及设备 Download PDF

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Publication number
WO2016202147A1
WO2016202147A1 PCT/CN2016/083042 CN2016083042W WO2016202147A1 WO 2016202147 A1 WO2016202147 A1 WO 2016202147A1 CN 2016083042 W CN2016083042 W CN 2016083042W WO 2016202147 A1 WO2016202147 A1 WO 2016202147A1
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Prior art keywords
board
user
digital
main control
control board
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PCT/CN2016/083042
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English (en)
French (fr)
Inventor
蓝雅燕
阮亮
朱爱锋
赵秋荷
孙小伟
徐柏山
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中兴通讯股份有限公司
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Publication of WO2016202147A1 publication Critical patent/WO2016202147A1/zh

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    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04M—TELEPHONIC COMMUNICATION
    • H04M7/00—Arrangements for interconnection between switching centres

Definitions

  • the present application relates to, but is not limited to, the field of communications, and in particular, to a method, device and device for processing a VoIP resource of a network telephone.
  • a relay is a transport call between two switching centers.
  • a trunk is a physical connection that carries multiple logical links.
  • Digital trunking refers to providing a digital physical channel and a digital trunk interface for the switch on the public telephone network, so that users can communicate with users on the public telephone network for voice and other information.
  • a digital relay can support 30 calls at the same time. In the related art, it is mainly a Primary Rate Interface (PRI) relay, also known as Primary Rate Access (PRA). Following.
  • PRI Primary Rate Interface
  • PRA Primary Rate Access
  • the NGN network carries all the services of the original PSTN network, offloads a large amount of data transmission to the IP network to alleviate the heavy load of the PSTN network, and adds and enhances many new and old services with the new features of the IP technology.
  • Voice over Internet Protocol digitizes analog signals and transmits them in real time over IP networks in the form of data packets.
  • the VOIP resources are centralized on the main control board, and the other narrow-band user boards that provide the access ports use the VOIP resources to implement the NGN voice service.
  • VOIP resources on the device may not be sufficient.
  • the number of call failures in the network element is increased, which has a great impact on the system VOIP resources. Reduce user experience satisfaction and even affect operator's operation management.
  • the embodiment of the invention provides a method, a device and a device for processing a VOIP resource of a network telephone, which solves the problem that the VOIP resource on the VOIP device in the related art causes the user to fail.
  • a method for processing a VoIP resource of a network phone comprising:
  • the digital relay board in the network element acquires user voice, and converts the user voice from an analog signal to a digital signal.
  • the digital relay board compresses the voice data stream corresponding to the digital signal to generate an IP voice packet.
  • the digital trunk board forwards the IP voice packet to an IP switching network of the main control board.
  • the method further includes: configuring, by the digital relay board, the following information:
  • the line ratio is a ratio of a number of trunk lines on the digital trunk board to a number of user lines on the digital trunk board,
  • the media IP and the port are used to identify the digital relay board.
  • a method for processing a VoIP resource of a network phone comprising:
  • the main control board in the network element records the hub ratio of the digital trunk board, and configures the media IP and port of the digital trunk board, wherein the digital trunk board is used to convert the user voice from the analog signal
  • the digital signal is compressed, and the voice data stream corresponding to the digital signal is compressed to generate an IP voice packet, and the IP voice packet is forwarded to the IP switching network of the main control board.
  • the main control board receives the access request of the user, and configures the digital relay accessed by the user according to the media IP, the port, the line ratio, and a preset configuration policy corresponding to the user. board.
  • the preset configuration policy includes one or more of the following:
  • the main control board preferentially configures the user to access the digital relay board when the user preferentially processes the user.
  • the main control board preferentially counts the number Accessing the user on the word trunk board;
  • the main control board does not access the user on the digital trunk board
  • the main control board selects a user who is not the relay board to access the digital relay board;
  • the main control board accesses the user of the relay board to another digital relay board.
  • the method further includes: forwarding, by the IP switching network of the main control board, the IP voice packet to the uplink board according to the port and IP information carried in the IP voice packet, or the master control The IP switching network of the board forwards the IP voice packet to the main control board according to the port and IP information carried by the IP voice packet.
  • a processing device for a VoIP resource of a network telephone comprising:
  • the conversion module is configured to acquire a user voice, and convert the user voice from an analog signal to a digital signal.
  • the generating module is configured to compress the voice data stream corresponding to the digital signal to generate an IP voice packet.
  • the first forwarding module is configured to forward the IP voice packet to the IP switching network of the main control board.
  • the device further includes: a configuration module, and a configuration module configured to: configure a media IP, a port, and a line ratio of the digital relay board, where the convergence ratio is The ratio of the number of trunks on the digital trunk board to the number of subscriber lines on the digital trunk board, the media IP and the port being used to identify the digital trunk board.
  • a processing device for a VoIP resource of a network telephone comprising:
  • a recording module configured to record a line ratio of the digital relay board and configured with a media IP and a port of the digital relay board, wherein the digital relay board is used to convert user voice from an analog signal to a digital And compressing the voice data stream corresponding to the digital signal to generate an IP voice packet, and forwarding the IP voice packet to the IP switching network of the main control board.
  • the receiving module is configured to receive an access request of the user, and configure the user to access according to the media IP, the port, the convergence ratio, and a preset configuration policy corresponding to the user.
  • Digital relay board configured to receive an access request of the user, and configure the user to access according to the media IP, the port, the convergence ratio, and a preset configuration policy corresponding to the user.
  • the preset configuration policy includes one or more of the following:
  • the main control board preferentially configures the user to access the digital relay board when the user preferentially processes the user.
  • the main control board preferentially accesses the user on the digital trunk board
  • the main control board does not access the user on the digital trunk board
  • the main control board selects a user who is not the relay board to access the digital relay board;
  • the main control board accesses the user of the relay board to another digital relay board.
  • the device further includes:
  • the second forwarding module is configured to forward the IP voice packet to the uplink board according to the port and IP information carried in the IP voice packet through the IP switching network of the main control board, or through the main control board
  • the IP switching network forwards the IP voice packet to the main control board according to the port and IP information carried by the IP voice packet.
  • a network telephone VOIP resource processing device includes a plurality of digital relay boards, and the data relay board includes: a processor and a VOIP module.
  • the VOIP module is configured to receive a digital signal that is converted by an analog signal of a user's voice.
  • the VOIP module is further configured to compress the voice data stream corresponding to the digital signal to generate an IP voice packet.
  • the processor is configured to forward the IP voice packet to an IP switching network of the main control board.
  • the main control board is configured to record a line ratio of the digital trunk board, and configure a media IP and a port of the digital trunk board; wherein the main control board is in the network element;
  • the main control board is further configured to receive an access request of the user, and according to the media IP, the The digital relay board accessed by the user is configured by the port, the line ratio, and a preset configuration policy corresponding to the user.
  • a computer readable storage medium storing computer executable instructions that, when executed by a processor, implement a method of processing the VOIP resources.
  • the digital relay board in the network element acquires the user voice, converts the user voice from an analog signal to a digital signal, and the digital relay board compresses the voice data stream corresponding to the digital signal to generate
  • the IP voice packet forwards the IP voice packet to the IP switching network of the main control board, solves the problem that the VOIP resource on the VOIP device is limited, causing the user to fail, and improves the stability of the VOIP device.
  • FIG. 1 is a flowchart 1 of a method for processing a VOIP resource of a network phone according to an embodiment of the present invention
  • FIG. 2 is a second flowchart of a method for processing a VoIP resource of a network phone according to an embodiment of the present invention
  • FIG. 3 is a structural block diagram 1 of a processing apparatus for a VoIP resource of a network telephone according to an embodiment of the present invention
  • FIG. 4 is a structural block diagram 2 of a processing apparatus for a VoIP resource of a network telephone according to an embodiment of the present invention
  • FIG. 5 is a schematic diagram of processing a network element distributed VOIP resource service according to an optional embodiment of the present invention.
  • FIG. 6 is a schematic diagram of physical configuration of a network element distributed VOIP resource according to an optional embodiment of the present invention.
  • FIG. 1 is a flowchart 1 of a method for processing a VOIP resource of a network telephone according to an embodiment of the present invention. As shown in FIG. S101-S103:
  • the digital relay board in the network element acquires user voice, and converts the user voice from an analog signal to a digital signal.
  • the digital relay board compresses the voice data stream corresponding to the digital signal to generate an IP voice packet.
  • the digital trunk board forwards the IP voice packet to the IP switching network of the main control board.
  • the digital relay board in the network element acquires the user voice, converts the user voice from the analog signal into a digital signal, and the digital relay board compresses the voice data stream corresponding to the digital signal to generate an IP voice packet.
  • the digital trunk board forwards the IP voice packet to the IP switching network of the main control board, which solves the problem that the VOIP resource on the VOIP device is limited, causing the user to fail the call, and improves the stability of the VOIP device.
  • the method further includes: configuring, by the digital relay board, the following information:
  • the line ratio is a ratio of the number of trunks on the digital trunk board to the number of subscriber lines on the digital trunk board, the media IP and the port Used to identify the digital relay board.
  • FIG. 2 is a second flowchart of a method for processing a VOIP resource of a network phone according to an embodiment of the present invention. As shown in FIG. 2, the process includes Steps S201-S202:
  • the main control board in the network element records a hub ratio of the digital trunk board, and configures a media IP and a port of the digital trunk board, where the digital trunk board is used to transmit user voice from an analog signal. Converting to a digital signal, and compressing the voice data stream corresponding to the digital signal, generating an IP voice packet, and forwarding the IP voice packet to the IP switching network of the main control board.
  • the main control board receives the access request of the user, and configures the digital relay board accessed by the user according to the media IP, the port, the line ratio, and a preset configuration policy corresponding to the user.
  • the main control board in the network element records the convergence ratio of the digital relay board, and configures a media IP and a port of the digital relay board, wherein the digital relay board converts user voice from an analog signal to a digital signal, and compresses the voice data stream corresponding to the digital signal to generate an IP voice packet, and And forwarding the IP voice packet to the IP switching network of the main control board, where the main control board receives the user's access request, and configures according to the media IP, the port, the line ratio, and a preset configuration policy corresponding to the user.
  • the digital trunk board accessed by the user solves the problem that the VOIP resource on the VOIP device is limited, causing the user to fail, and improves the stability of the VOIP device.
  • the preset configuration policy includes one or more of the following:
  • the main control board preferentially configures the user to access the digital relay board
  • the main control board preferentially accesses the user on the digital trunk board
  • the main control board does not access the user on the digital trunk board
  • the main control board selects a user who is not the relay board to access the digital relay board;
  • the main control board accesses the user of the relay board to another digital trunk board.
  • the method further includes: the IP switching network of the main control board forwards the IP voice packet to the uplink board according to the port and IP information carried in the IP voice packet, or the IP of the main control board The switching network forwards the IP voice packet to the main control board according to the port and IP information carried by the IP voice packet.
  • a processing device for the VoIP resource of the VoIP is provided, and the device is used to implement the foregoing embodiments and optional implementations, and details are not described herein.
  • the term "module” may implement a combination of software and/or hardware of a predetermined function.
  • the apparatus described in the following embodiments is preferably implemented in software, hardware, or a combination of software and hardware, is also possible and contemplated.
  • FIG. 3 is a structural block diagram 1 of a processing apparatus for a VoIP resource of a network telephone according to an embodiment of the present invention.
  • the apparatus may be applied to a digital trunk board in a network element, as shown in FIG. include:
  • the conversion module 31 is configured to acquire a user voice and convert the user voice from an analog signal to a digital signal.
  • the generating module 32 is configured to compress the voice data stream corresponding to the digital signal to generate an IP voice packet.
  • the first forwarding module 33 is configured to forward the IP voice packet to the IP switching network of the main control board.
  • the conversion module 31 acquires the user voice, converts the user voice from the analog signal into a digital signal, and the generating module 32 compresses the voice data stream corresponding to the digital signal to generate an IP voice packet, and the first forwarding module 33 is The IP switching network of the control board forwards the IP voice packet, which solves the problem that the VOIP resource on the VOIP device is limited, causing the user to fail, and improves the stability of the VOIP device.
  • the device further includes: a configuration module, and a configuration module configured to: configure a media IP, a port, and a line ratio of the digital relay board, where the convergence ratio is the digital relay The ratio of the number of trunks on the board to the number of subscriber lines on the digital trunk board, the media IP and the port being used to identify the digital trunk board.
  • a configuration module configured to: configure a media IP, a port, and a line ratio of the digital relay board, where the convergence ratio is the digital relay The ratio of the number of trunks on the board to the number of subscriber lines on the digital trunk board, the media IP and the port being used to identify the digital trunk board.
  • FIG. 4 is a structural block diagram 2 of a processing apparatus for a VoIP resource of a network telephone according to an embodiment of the present invention.
  • the apparatus may be applied to a main control board in a network element, as shown in FIG. 4, where the apparatus includes
  • the recording module 41 is configured to record a line ratio of the digital relay board and configured with a media IP and a port of the digital relay board, wherein the digital relay board is used to convert user voice from an analog signal to a digital signal And compressing the voice data stream corresponding to the digital signal, generating an IP voice packet, and forwarding the IP voice packet to the IP switching network of the main control board;
  • the receiving module 42 is configured to receive an access request of the user, and configure the digital relay board accessed by the user according to the media IP, the port, the line ratio, and a preset configuration policy corresponding to the user.
  • the recording module 41 on the main control board in the network element records the line ratio of the digital relay board, and is configured with the media IP and port of the digital relay board, wherein the digital relay board Converting the user voice from an analog signal to a digital signal, compressing the voice data stream corresponding to the digital signal, generating an IP voice packet, and forwarding the IP voice packet to the IP switching network of the main control board,
  • the receiving module 42 receives the access request of the user, and configures the digital relay board accessed by the user according to the media IP, the port, the line ratio, and the preset configuration policy corresponding to the user, and solves the problem on the VOIP device.
  • the limited VOIP resources cause the user to fail the call and improve the stability of the VOIP device.
  • the preset configuration policy includes one or more of the following:
  • the main control board preferentially configures the user to access the digital relay board
  • the main control board preferentially accesses the user on the digital trunk board
  • the main control board does not access the user on the digital trunk board
  • the main control board selects a user who is not the relay board to access the digital relay board;
  • the main control board accesses the user of the relay board to another digital trunk board.
  • the device further includes a second forwarding module 43 configured to forward the IP voice packet to the uplink board according to the port and IP information carried in the IP voice packet through the IP switching network of the main control board, or The IP switching network of the main control board forwards the IP voice packet to the main control board according to the port and IP information carried by the IP voice packet.
  • a second forwarding module 43 configured to forward the IP voice packet to the uplink board according to the port and IP information carried in the IP voice packet through the IP switching network of the main control board, or The IP switching network of the main control board forwards the IP voice packet to the main control board according to the port and IP information carried by the IP voice packet.
  • a network telephone VOIP resource processing device which includes a plurality of digital relay boards, and the data relay board includes: a processor and a VOIP module.
  • the VOIP module is configured to receive a digital signal that is generated by an analog signal conversion of a user's voice.
  • the VOIP module is further configured to compress the voice data stream corresponding to the digital signal to generate an IP voice packet.
  • the processor is configured to forward the IP voice packet to an IP switching network of the main control board.
  • the optional embodiment provides a distributed VOIP resource sharing method based on the relay.
  • the number of PRI users in the network element is large, the impact and consumption on the system VOIP voice resources can be reduced, and the system convergence ratio is sharply increased. The situation improves system performance and engineering support capabilities.
  • the optional embodiment provides a method for distributed VOIP resource sharing based on digital relay.
  • the network element of the distributed VOIP based on relay is physically added with multiple digital trunk boards with VOIP resources on the network element.
  • a separate CPU chip and VOIP resource physical bearer are configured on the digital trunk board.
  • the digital trunk board can not only be used as a PRI access, but also can provide VOIP resources for the PRI user or other users of the system, and implement VOIP load sharing on the network element.
  • FIG. 5 is a schematic diagram of a distributed VOIP resource service processing of a network element according to a preferred embodiment of the present invention. As shown in FIG. 5, the internal processing flow is to perform a voice service of a PRI user on a digital trunk board when a PRI user performs an NGN voice service.
  • This board is IP.
  • the IP of this board includes:
  • PRI user voice is transmitted to the board through the physical channel as an analog signal.
  • the internal signal of the board is converted into a digital signal and transmitted to the VOIP module of the board, and the voice data stream is compressed and packaged in the VOIP module.
  • the generated IP voice packet is forwarded to the IP switching network of the main control board through the CPU of the board.
  • the IP switching network forwards the packet to the uplink board according to the port and IP information carried in the IP voice packet, or forwards it to the CPU of the main control board (when the PRI user talks with the internal user of the network element).
  • the network element of the distributed VOIP based on the relay needs to centrally manage all the VOIP resources on the main control board.
  • the VOIP on the main control board is called the system VOIP.
  • the VOIP resource data table record is added to record the physical location information of each VOIP resource in the current network element, and the actual number of VOIP resources that can be provided.
  • VOIP resource pool that is, a collection of VOIP resources, and manage the state and usage of VOIP resources in the resource pool with certain rules.
  • the VOIP resource selection order and the line ratio are different to meet the VOIP resources of different user levels. For example, a high-level user can satisfy an instant voice call, and a low-level user can You need to wait in line when the system is busy.
  • the default selection order is as follows:
  • the VOIP resources of the board are preferentially selected.
  • the VOIP resources of the system are preferentially selected.
  • the VOIP resources of the system are all occupied, if there are remaining VOIP resources on the digital trunk board, the VOIP resources on the digital trunk board can be selected.
  • the VOIP resources reserved for the users of the digital trunk board are calculated according to the set line ratio, and the reserved amount cannot exceed the maximum number of VOIP resources that can be actually provided on the digital trunk board;
  • the VOIP resources of the system are preferentially selected, and the remaining VOIP resources on the digital trunk board are selected.
  • Optional distributed VOIP resource mode to plan and configure the VOIP resource distribution of network elements.
  • each media VoIP resource and port for each VOIP resource including system VOIP and VOIP on the digital trunk board, to identify it in the network element global IP switching network.
  • the optional embodiment has simple configuration, resource distribution, balanced load, and high reliability.
  • the VOIP resources are distributed on the digital trunk board to reduce the consumption of the system VOIP resources by the digital relay PRI users. On the basis of ensuring that the PRI users of the board have sufficient VOIP resources, the remaining VOIP resources can be shared at the system level. In order to improve system performance, reduce costs and improve engineering support capabilities.
  • FIG. 6 is a schematic diagram of physical configuration of a distributed VOIP resource of a network element according to an alternative embodiment of the present invention.
  • a network element may be distributed with n (n is less than or equal to the slot in the network element).
  • the total number is -6, and there are two power boards, two uplink boards, and two main control boards for the active/standby protection mechanism.
  • the digital trunking board can be configured with other service boards.
  • Step 1 Configure the corresponding board.
  • Step 2 Configure the media IP and port for each digital trunk.
  • Step 3 Select the line ratio of each digital trunk board. The default is 0, that is, the VOIP resources in the whole network element can be mixed.
  • the digital hop PRI user When the number of PRI users of the digital trunk is greater than the VOIP resources provided by the system master, the digital hop PRI user does not consume the system VOIP resources.
  • the other service board users and the digital relay PRI users can perform NGN voice normally. business.
  • the NGN voice service of the digital trunk PRI user is normal.
  • the other service board users can automatically select the VOIP resources on the digital trunk board, and the NGN voice service is normal, which improves the engineering support. ability.
  • the distributed VOIP resource sharing method based on relay is more advantageous than the traditional centralized VOIP resource sharing or pure distributed VOIP resource sharing method, which is simple and easy to use, saves cost, improves engineering support capability, and can be widely applied to railways. Governments, institutions, and national operators such as highways, military, and mines.
  • a computer readable storage medium storing computer executable instructions that, when executed by a processor, implement a method of processing the VOIP resources.
  • the embodiment of the invention further provides a storage medium.
  • the storage medium may be configured to store program code for performing the method steps of the above embodiment:
  • the foregoing storage medium may include, but not limited to, a USB flash drive, a Read-Only Memory (ROM), a Random Access Memory (RAM), a mobile hard disk, and a magnetic memory.
  • ROM Read-Only Memory
  • RAM Random Access Memory
  • a mobile hard disk e.g., a hard disk
  • magnetic memory e.g., a hard disk
  • the processor performs the method steps of the foregoing embodiments according to the stored program code in the storage medium.
  • each module or step of the above-described embodiments of the present invention can be implemented by a general-purpose computing device, which can be centralized on a single computing device or distributed across multiple computing devices. Alternatively, they may be implemented by program code executable by the computing device such that they may be stored in the storage device by the computing device and, in some cases, may be different.
  • the steps shown or described herein are performed sequentially, or they are separately fabricated into a plurality of integrated circuit modules, or a plurality of the modules or steps are fabricated into a single integrated circuit module.
  • embodiments of the invention are not limited to any specific combination of hardware and software.
  • all or part of the steps of the above embodiments may also be implemented by using an integrated circuit. These steps may be separately fabricated into individual integrated circuit modules, or multiple modules or steps may be fabricated into a single integrated circuit module. achieve.
  • the devices/function modules/functional units in the above embodiments may be implemented by a general-purpose computing device, which may be centralized on a single computing device or distributed over a network of multiple computing devices.
  • the device/function module/functional unit in the above embodiment When the device/function module/functional unit in the above embodiment is implemented in the form of a software function module and sold or used as a stand-alone product, it can be stored in a computer readable storage medium.
  • the above mentioned computer readable storage medium may be a read only memory, a magnetic disk or an optical disk or the like.
  • the digital relay board in the network element acquires the user voice, converts the user voice from an analog signal to a digital signal, and the digital relay board compresses the voice data stream corresponding to the digital signal to generate
  • the IP voice packet forwards the IP voice packet to the IP switching network of the main control board, solves the problem that the VOIP resource on the VOIP device is limited, causing the user to fail, and improves the stability of the VOIP device.

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Abstract

本申请提供了一种网络电话VOIP资源处理方法、装置及设备，其中，该方法通过网元中的数字中继板获取用户语音，将该用户语音从模拟信号转换为数字信号，该数字中继板对该数字信号对应的语音数据流进行压缩，生成IP语音包，该数字中继板向主控板的IP交换网转发该IP语音包。

Description

一种网络电话VOIP资源处理方法、装置及设备 技术领域
本申请涉及但不限于通信领域，尤其涉及一种网络电话VOIP资源处理方法、装置及设备。
背景技术
中继(Relay)是两个交换中心之间的一条传输通话，中继线是承载多条逻辑链路的一条物理连接。数字中继是指在公众电话网上，提供一个数字物理通道和交换机数字中继接口，使用户与公众电话网上的用户进行语音等信息的互通。一路数字中继能同时支持30路通话，在相关技术中，主要是基群速率接口(Primary Rate Interface，简称PRI)中继，也称为基群速率接入(Primary Rate Access，简称PRA)中继。
随着科技的发展，从传统的以电路交换为主的公共交换电话网络(Public Switched Telephone Network，简称PSTN)，逐渐迈向以分组交换为主的下一代网络(Next Generation Network，简称NGN)。NGN网络承载了原有PSTN网络的所有业务，把大量的数据传输卸载到IP网络中以减轻PSTN网络的重荷，又以IP技术的新特性增加和增强了许多新老业务。
而网络电话(Voice over Internet Protocol，简称VOIP)就是将模拟信号数字化，以数据封包的形式在IP网络上做实时传递。
在相关技术中的集中式VOIP系统，将VOIP资源集中在主控板上，而其他提供接入端口的窄带用户板共同使用其VOIP资源，实现接入用户提供NGN语音业务。
在VOIP系统中，当网元的PRI中继用户足够多时，会出现设备上的VOIP资源不够用的情况，表现为网元中的用户呼叫失败次数攀升，对系统VOIP资源产生很大的冲击，降低用户体验满意度，甚至影响运营商的运营管理。
针对相关技术中，VOIP设备上的VOIP资源有限导致用户呼叫失败的问题，目前尚无提出有效的解决方案。
发明内容
以下是对本文详细描述的主题的概述。本概述并非是为了限制权利要求的保护范围。
本发明实施例提供了一种网络电话VOIP资源处理方法、装置及设备，解决了相关技术中VOIP设备上的VOIP资源有限导致用户呼叫失败的问题。
一种网络电话VOIP资源的处理方法，包括：
网元中的数字中继板获取用户语音，将所述用户语音从模拟信号转换为数字信号。
所述数字中继板对所述数字信号对应的语音数据流进行压缩，生成IP语音包。
所述数字中继板向主控板的IP交换网转发所述IP语音包。
可选地，所述方法还包括：所述数字中继板进行以下信息的配置：
所述数字中继板的媒体IP、端口及集线比，其中，所述集线比为所述数字中继板上的中继线数与所述数字中继板上用户线数的比值，所述媒体IP和所述端口用于对所述数字中继板进行识别。
一种网络电话VOIP资源的处理方法，包括：
在网元中的主控板记录数字中继板的集线比，并配置所述数字中继板的媒体IP和端口，其中，所述数字中继板板用于将用户语音从模拟信号转换为数字信号，并对所述数字信号对应的语音数据流进行压缩，生成IP语音包，向主控板的IP交换网转发所述IP语音包。
所述主控板接收用户的接入请求，并根据所述媒体IP、所述端口、所述集线比以及与所述用户对应的预设配置策略，配置所述用户接入的数字中继板。
可选地，所述预设配置策略包括以下一种或多种：
在所述用户为预设优先处理用户的情况下，所述主控板优先给所述用户配置接入的数字中继板；
在所述用户为数字中继板上的用户的情况下，所述主控板优先在所述数 字中继板上接入所述用户；
在所述用户不是数字中继板上的用户，且所述数字中继板上不存在空闲线路的情况下，所述主控板不在所述数字中继板上接入所述用户；
在所述数字中继板上存在空闲线路的情况下，所述主控板选取不是所述中继板的用户接入所述数字中继板；以及，
在所述数字中继板上为用户配置的线路被全部占用的情况下，所述主控板将所述中继板的用户接入其他的数字中继板。
可选地，所述方法还包括：所述主控板的IP交换网根据所述IP语音包中携带的端口和IP信息将所述IP语音包转发给上联板，或者，所述主控板的IP交换网根据所述IP语音包携带的端口和IP信息将所述IP语音包转发给所述主控板。
一种网络电话VOIP资源的处理装置，包括：
转换模块，设置为获取用户语音，将所述用户语音从模拟信号转换为数字信号。
生成模块，设置为对所述数字信号对应的语音数据流进行压缩，生成IP语音包。
第一转发模块，设置为向主控板的IP交换网转发所述IP语音包。
可选地，所述装置还包括：配置模块；配置模块，设置为进行以下信息的配置：配置所述数字中继板的媒体IP、端口及集线比，其中，所述集线比为所述数字中继板上的中继线数与所述数字中继板上用户线数的比值，所述媒体IP和所述端口用于对所述数字中继板进行识别。
一种网络电话VOIP资源的处理装置，包括：
记录模块，设置为记录数字中继板的集线比，并配置有所述数字中继板的媒体IP和端口，其中，所述数字中继板板用于将用户语音从模拟信号转换为数字信号，并对所述数字信号对应的语音数据流进行压缩，生成IP语音包，向主控板的IP交换网转发所述IP语音包。
接收模块，设置为接收用户的接入请求，并根据所述媒体IP、所述端口、所述集线比以及与所述用户对应的预设配置策略，配置所述用户接入的 数字中继板。
可选地，所述预设配置策略包括以下一种或多种：
在所述用户为预设优先处理用户的情况下，所述主控板优先给所述用户配置接入的数字中继板；
在所述用户为数字中继板上的用户的情况下，所述主控板优先在所述数字中继板上接入所述用户；
在所述用户不是数字中继板上的用户，且所述数字中继板上不存在空闲线路的情况下，所述主控板不在所述数字中继板上接入所述用户；
在所述数字中继板上存在空闲线路的情况下，所述主控板选取不是所述中继板的用户接入所述数字中继板；以及，
在所述数字中继板上为用户配置的线路被全部占用的情况下，所述主控板将所述中继板的用户接入其他的数字中继板。
可选地，所述装置还包括：
第二转发模块，设置为通过所述主控板的IP交换网根据所述IP语音包中携带的端口和IP信息将所述IP语音包转发给上联板，或者，通过所述主控板的IP交换网根据所述IP语音包携带的端口和IP信息将所述IP语音包转发给所述主控板。
一种网络电话VOIP资源处理设备，包括多个数字中继板，所述数据中继板上包括：处理器和VOIP模块。
所述VOIP模块，设置为接收数字信号，所述数字信号是由用户语音的模拟信号转换成的。
所述VOIP模块，还设置为对所述数字信号对应的语音数据流进行压缩，生成IP语音包。
所述处理器，设置为将所述IP语音包转发至主控板的IP交换网。
所述主控板，设置为记录数字中继板的集线比，并配置所述数字中继板的媒体IP和端口；其中，所述主控板处于网元中；
所述主控板，还设置为接收用户的接入请求，并根据所述媒体IP、所述 端口、所述集线比以及与所述用户对应的预设配置策略，配置所述用户接入的数字中继板。
一种计算机可读存储介质，存储有计算机可执行指令，所述计算机可执行指令被处理器执行时实现所述的VOIP资源的处理方法。
通过本发明实施例的方案，网元中的数字中继板获取用户语音，将该用户语音从模拟信号转换为数字信号，该数字中继板对该数字信号对应的语音数据流进行压缩，生成IP语音包，该数字中继板向主控板的IP交换网转发该IP语音包，解决了VOIP设备上的VOIP资源有限导致用户呼叫失败的问题，提高了VOIP设备的稳定性。
附图概述
图1是根据本发明实施例的一种网络电话VOIP资源的处理方法的流程图一；
图2是根据本发明实施例的一种网络电话VOIP资源的处理方法的流程图二；
图3是根据本发明实施例的一种网络电话VOIP资源的处理装置的结构框图一；
图4是根据本发明实施例的一种网络电话VOIP资源的处理装置的结构框图二；
图5是根据本发明可选实施例的网元分布式VOIP资源业务处理示意图；
图6是根据本发明可选实施例的网元分布式VOIP资源物理配置示意图。
本发明的实施方式
下文中将参考附图并结合实施例进行详细描述。需要说明的是，在不冲突的情况下，本申请中的实施例及实施例中的特征可以相互组合。
需要说明的是，本发明实施例的说明书和权利要求书及上述附图中的术语“第一”、“第二”等是用于区别类似的对象，而不必用于描述特定的顺序或 先后次序。
在本实施例中提供了一种网络电话VOIP资源的处理方法，图1是根据本发明实施例的一种网络电话VOIP资源的处理方法的流程图一，如图1所示，该流程包括步骤S101-S103：
S101、网元中的数字中继板获取用户语音，将该用户语音从模拟信号转换为数字信号。
S102、该数字中继板对该数字信号对应的语音数据流进行压缩，生成IP语音包。
S103、该数字中继板向主控板的IP交换网转发该IP语音包。
通过上述步骤，网元中的数字中继板获取用户语音，将该用户语音从模拟信号转换为数字信号，该数字中继板对该数字信号对应的语音数据流进行压缩，生成IP语音包，该数字中继板向主控板的IP交换网转发该IP语音包，解决了VOIP设备上的VOIP资源有限导致用户呼叫失败的问题，提高了VOIP设备的稳定性。
可选地，所述方法还包括：所述数字中继板进行以下信息的配置：
该数字中继板的媒体IP、端口及集线比，其中，该集线比为该数字中继板上的中继线数与该数字中继板上用户线数的比值，该媒体IP和该端口用于对该数字中继板进行识别。
在本实施例中还提供了一种网络电话VOIP资源的处理方法，图2是根据本发明实施例的一种网络电话VOIP资源的处理方法的流程图二，如图2所示，该流程包括步骤S201-S202：
S201、在该网元中的主控板记录数字中继板的集线比，并配置该数字中继板的媒体IP和端口，其中，该数字中继板板用于将用户语音从模拟信号转换为数字信号，并对该数字信号对应的语音数据流进行压缩，生成IP语音包，向主控板的IP交换网转发该IP语音包。
S202、该主控板接收用户的接入请求，并根据该媒体IP、该端口、该集线比以及与该用户对应的预设配置策略，配置该用户接入的数字中继板。
通过上述步骤，在该网元中的主控板记录数字中继板的集线比，并配置 有该数字中继板的媒体IP和端口，其中，该数字中继板板将用户语音从模拟信号转换为数字信号，并对该数字信号对应的语音数据流进行压缩，生成IP语音包，以及向主控板的IP交换网转发该IP语音包，该主控板接收用户的接入请求，并根据该媒体IP、该端口、该集线比以及与该用户对应的预设配置策略，配置该用户接入的数字中继板，解决了VOIP设备上的VOIP资源有限导致用户呼叫失败的问题，提高了VOIP设备的稳定性。
可选地，该预设配置策略包括以下一种或多种：
在该用户为预设优先处理用户的情况下，该主控板优先给该用户配置接入的数字中继板；
在该用户为数字中继板上的用户的情况下，该主控板优先在该数字中继板上接入该用户；
在该用户不是数字中继板上的用户，且该数字中继板上不存在空闲线路的情况下，该主控板不在该数字中继板上接入该用户；
在该数字中继板上存在空闲线路的情况下，该主控板选取不是该中继板的用户接入该数字中继板；以及，
在该数字中继板上为用户配置的线路被全部占用的情况下，该主控板将该中继板的用户接入其他的数字中继板。
可选地，所述方法还包括：该主控板的IP交换网根据该IP语音包中携带的端口和IP信息将所述IP语音包转发给上联板，或者，该主控板的IP交换网根据该IP语音包携带的端口和IP信息将所述IP语音包转发给该主控板。
在本实施例中还提供了一种网络电话VOIP资源的处理装置，该装置用于实现上述实施例及可选实施方式，已经进行过说明的不再赘述。如以下所使用的，术语“模块”可以实现预定功能的软件和/或硬件的组合。尽管以下实施例所描述的装置较佳地以软件来实现，但是硬件，或者软件和硬件的组合的实现也是可能并被构想的。
图3是根据本发明实施例的一种网络电话VOIP资源的处理装置的结构框图一，该装置可以应用于网元中的数字中继板中，如图3所示，该装置包 括：
转换模块31，设置为获取用户语音，将该用户语音从模拟信号转换为数字信号。
生成模块32，设置为对该数字信号对应的语音数据流进行压缩，生成IP语音包。
第一转发模块33，设置为向主控板的IP交换网转发该IP语音包。
通过上述装置，转换模块31获取用户语音，将该用户语音从模拟信号转换为数字信号，生成模块32对该数字信号对应的语音数据流进行压缩，生成IP语音包，第一转发模块33向主控板的IP交换网转发该IP语音包，解决了VOIP设备上的VOIP资源有限导致用户呼叫失败的问题，提高了VOIP设备的稳定性。
可选地，该装置还包括：配置模块；配置模块，设置为进行以下信息的配置：配置该数字中继板的媒体IP、端口及集线比，其中，该集线比为该数字中继板上的中继线数与该数字中继板上用户线数的比值，该媒体IP和该端口用于对该数字中继板进行识别。
图4是根据本发明实施例的一种网络电话VOIP资源的处理装置的结构框图二，该装置可以应用于网元中的主控板中，如4所示，该装置包括
记录模块41，设置为记录数字中继板的集线比，并配置有该数字中继板的媒体IP和端口，其中，该数字中继板板用于将用户语音从模拟信号转换为数字信号，并对该数字信号对应的语音数据流进行压缩，生成IP语音包，向主控板的IP交换网转发该IP语音包；
接收模块42，设置为接收用户的接入请求，并根据该媒体IP、该端口、该集线比以及与该用户对应的预设配置策略，配置该用户接入的数字中继板。
通过上述装置，在该网元中的主控板上的记录模块41记录数字中继板的集线比，并配置有该数字中继板的媒体IP和端口，其中，该数字中继板板将用户语音从模拟信号转换为数字信号，并对该数字信号对应的语音数据流进行压缩，生成IP语音包，以及向主控板的IP交换网转发该IP语音包，接 收模块42接收用户的接入请求，并根据该媒体IP、该端口、该集线比以及与该用户对应的预设配置策略，配置该用户接入的数字中继板，解决了VOIP设备上的VOIP资源有限导致用户呼叫失败的问题，提高了VOIP设备的稳定性。
可选地，该预设配置策略包括以下一种或多种：
在该用户为预设优先处理用户的情况下，该主控板优先给该用户配置接入的数字中继板；
在该用户为数字中继板上的用户的情况下，该主控板优先在该数字中继板上接入该用户；
在该用户不是数字中继板上的用户的情况下，，且所述数字中继板上不存在空闲线路该主控板不在该数字中继板上接入该用户；
在该数字中继板上存在空闲线路的情况下，该主控板选取不是该中继板的用户接入该数字中继板；以及，
在该数字中继板上为用户配置的线路被全部占用的情况下，该主控板将该中继板的用户接入其他的数字中继板。
可选地，该装置还包括第二转发模块43，设置为通过该主控板的IP交换网根据该IP语音包中携带的端口和IP信息将所述IP语音包转发给上联板，或者，该主控板的IP交换网根据该IP语音包携带的端口和IP信息将所述IP语音包转发给该主控板。
在本发明实施例中，还提供了一种网络电话VOIP资源处理设备，包括多个数字中继板，该数据中继板上包括：处理器和VOIP模块。
该VOIP模块，设置为接收数字信号，该数字信号是由用户语音的模拟信号转换生成的。
该VOIP模块，还设置为对该数字信号对应的语音数据流进行压缩，生成IP语音包。
该处理器，设置为将该IP语音包转发至主控板的IP交换网。
下面结合可选实施例和实施方式进行说明。
本可选实施例提供了一种基于中继的分布式VOIP资源共享方法，当网元中PRI用户数量较多时，能够减少对系统VOIP语音资源产生的冲击和消耗，降低系统集线比陡升的情况，提升系统性能和工程保障能力。
本可选实施例提供一种基于数字中继的分布式VOIP资源共享的方法。基于中继的分布式VOIP的网元，物理上的表现就是在网元上增加多块带有VOIP资源的数字中继板。数字中继板上配置单独的CPU芯片及VOIP资源物理承载。这样数字中继板上不仅可以作为PRI接入，也可以为本板PRI用户或系统其它用户提供VOIP资源，在网元上实现VOIP的负荷分担。
图5是根据本发明优选实施例的网元分布式VOIP资源业务处理示意图，如图5所示，内部处理流程是在PRI用户进行NGN语音业务时，将数字中继板上PRI用户的语音业务本板IP化。本板IP化包括：
1.PRI用户语音以模拟信号通过物理通道传输到单板。
2.单板内部将模拟信号转化为数字信号传输到本板的VOIP模块中，在VOIP模块内对语音数据流进行压缩打包。
3.产生的IP语音包，经过本板CPU转发到主控板的IP交换网。
4.IP交换网根据IP语音包携带的端口和IP信息，转发到上联板发送出去，或是转发到主控板的CPU处理(当PRI用户与网元内部用户通话时)。
基于中继的分布式VOIP的网元，要在主控板上对所有的VOIP资源集中管理。为了方便理解，将主控板上的VOIP称为系统VOIP。
一、将分布的VOIP资源集中到主控板上统一管理。
在系统的数据库中，增加VOIP资源数据表记录，以记录当前网元中各个VOIP资源的物理位置信息、实际可以提供的VOIP资源数量等。
并创建一个VOIP资源池，即VOIP资源的集合，以一定规则管理资源池中VOIP资源的状态和使用情况。
二、不同级别的用户依据不同的规则选取VOIP资源。
根据用户的级别不同，对应不同的VOIP资源选取顺序和集线比(中继线数：用户线数)，以满足不同用户级别的VOIP资源满足，如高级别用户可以满足即时语音通话，低级别用户在系统繁忙时需要排队等待等。
默认的选择顺序如下：
1.对于数字中继板上的用户，优先选取本板的VOIP资源。
2.对于网元中其他单板上的用户，优先选取系统的VOIP资源。
3.数字中继板上的用户，如果本板的VOIP资源已全部被占用，则选取系统的VOIP资源；
4.网元中其他的用户，如果系统的VOIP资源已全部被占用，在数字中继板上VOIP资源还有剩余的情况下，可以选取数字中继板上的VOIP资源。
在设置了集线比的情况下，根据集线比计算出预留给数字中继板用户使用的VOIP资源，而预留的数量不能超过数字中继板上实际能提供VOIP资源的最大数量；
对于同网元其他单板用户，优先选取系统的VOIP资源，其次是选取数字中继板上剩下的VOIP资源。
在人机交互界面和网管上都有一套相应的配置命令。详细操作为：
1.选配分布式VOIP资源模式，对网元VOIP资源分布进行规划配置。
2.为每个VOIP资源配置其媒体IP和端口，包括系统VOIP及数字中继板上VOIP，以便在网元全局IP交换网中对其进行识别。
3.选配数字中继用户的集线比。
基于中继的分布式VOIP资源系统一旦配置建立，在承载E1故障时、主备倒换、主控板或某一块中继板上VOIP故障时，可以自动切换使用其他正常的VOIP资源，自动实现保护功能，提升工程保障能力。
本可选实施例配置简单、资源分布、均衡负载、可靠性高。
将VOIP资源分布在数字中继板上，以减少数字中继PRI用户对系统VOIP资源的消耗，在保证本板PRI用户有足够的VOIP资源使用的基础上，剩余的VOIP资源可以系统级别共享，以便提升系统性能、降低成本并提升工程保障能力。
在本实施例中，图6是根据本发明可选实施例的网元分布式VOIP资源物理配置示意图，如图6所示，一个网元内可分布n(n小于等于网元内槽位 总数-6，其中有2块电源板、2块上联板、2块主控板用于主备保护机制)块数字中继板，其余空闲的槽位可配置其他业务单板。
本可选实施例的配置方法步骤包括步骤1-3：
步骤1.配置相应的单板。
步骤2.配置每个数字中继板的媒体IP和端口。
步骤3.选配每个数字中继板的集线比，默认为0，即全网元内的VOIP资源可以混用。
在数字中继PRI用户数大于系统主控提供的VOIP资源时，并未出现数字中继PRI用户对系统VOIP资源的消耗，其他业务单板用户和各个数字中继PRI用户都能正常进行NGN语音业务。
主备倒换时，正在进行中的NGN语音业务丝毫没有收到影响。
在系统VOIP资源出现故障时，数字中继PRI用户正在进行中的NGN语音业务正常，其他业务单板用户可以自动选取数字中继板上的VOIP资源，且其NGN语音业务正常，提升了工程保障能力。
基于中继的分布式VOIP资源共享方法比传统的集中式VOIP资源共享或单纯的分布式VOIP资源共享方法更具有优越性，简单易用，节约成本，提升工程保障能力，并且可以广泛应用于铁路、公路、军队、矿井等政府企事业单位及各国运营商。
一种计算机可读存储介质，存储有计算机可执行指令，所述计算机可执行指令被处理器执行时实现所述的VOIP资源的处理方法。
本发明实施例还提供了一种存储介质。可选地，在本实施例中，上述存储介质可以被设置为存储用于执行上述实施例的方法步骤的程序代码：
可选地，在本实施例中，上述存储介质可以包括但不限于：U盘、只读存储器(ROM，Read-Only Memory)、随机存取存储器(RAM，Random Access Memory)、移动硬盘、磁碟或者光盘等各种可以存储程序代码的介质。
可选地，在本实施例中，处理器根据存储介质中已存储的程序代码执行上述实施例的方法步骤。
显然，本领域的技术人员应该明白，上述的本发明实施例的每个模块或每个步骤可以用通用的计算装置来实现，它们可以集中在单个的计算装置上，或者分布在多个计算装置所组成的网络上，可选地，它们可以用计算装置可执行的程序代码来实现，从而，可以将它们存储在存储装置中由计算装置来执行，并且在某些情况下，可以以不同于此处的顺序执行所示出或描述的步骤，或者将它们分别制作成多个集成电路模块，或者将它们中的多个模块或步骤制作成单个集成电路模块来实现。这样，本发明实施例不限制于任何特定的硬件和软件结合。
本领域普通技术人员可以理解上述实施例的全部或部分步骤可以使用计算机程序流程来实现，所述计算机程序可以存储于一计算机可读存储介质中，所述计算机程序在相应的硬件平台上(如系统、设备、装置、器件等)执行，在执行时，包括方法实施例的步骤之一或其组合。
可选地，上述实施例的全部或部分步骤也可以使用集成电路来实现，这些步骤可以被分别制作成一个个集成电路模块，或者将它们中的多个模块或步骤制作成单个集成电路模块来实现。
上述实施例中的装置/功能模块/功能单元可以采用通用的计算装置来实现，它们可以集中在单个的计算装置上，也可以分布在多个计算装置所组成的网络上。
上述实施例中的装置/功能模块/功能单元以软件功能模块的形式实现并作为独立的产品销售或使用时，可以存储在一个计算机可读取存储介质中。上述提到的计算机可读取存储介质可以是只读存储器，磁盘或光盘等。
工业实用性
通过本发明实施例的方案，网元中的数字中继板获取用户语音，将该用户语音从模拟信号转换为数字信号，该数字中继板对该数字信号对应的语音数据流进行压缩，生成IP语音包，该数字中继板向主控板的IP交换网转发该IP语音包，解决了VOIP设备上的VOIP资源有限导致用户呼叫失败的问题，提高了VOIP设备的稳定性。

Claims (12)

  1. 一种网络电话VOIP资源的处理方法，包括：
    网元中的数字中继板获取用户语音，将所述用户语音从模拟信号转换为数字信号；
    所述数字中继板对所述数字信号对应的语音数据流进行压缩，生成IP语音包；
    所述数字中继板向主控板的IP交换网转发所述IP语音包。
  2. 根据权利要求1所述的VOIP资源的处理方法，所述方法还包括：所述数字中继板进行以下信息的配置：
    所述数字中继板的媒体IP、端口及集线比，其中，所述集线比为所述数字中继板上的中继线数与所述数字中继板上用户线数的比值，所述媒体IP和所述端口用于对所述数字中继板进行识别。
  3. 一种网络电话VOIP资源的处理方法，包括：
    在网元中的主控板记录数字中继板的集线比，并配置所述数字中继板的媒体IP和端口，其中，所述数字中继板板用于将用户语音从模拟信号转换为数字信号，并对所述数字信号对应的语音数据流进行压缩，生成IP语音包，向主控板的IP交换网转发所述IP语音包；
    所述主控板接收用户的接入请求，并根据所述媒体IP、所述端口、所述集线比以及与所述用户对应的预设配置策略，配置所述用户接入的数字中继板。
  4. 根据权利要求3所述的VOIP资源的处理方法，其中，所述预设配置策略包括以下一种或多种：
    在所述用户为预设优先处理用户的情况下，所述主控板优先给所述用户配置接入的数字中继板；
    在所述用户为数字中继板上的用户的情况下，所述主控板优先在所述数字中继板上接入所述用户；
    在所述数字中继板上存在空闲线路的情况下，所述主控板选取不是所述 中继板的用户接入所述数字中继板；以及，
    在所述数字中继板上为用户配置的线路被全部占用的情况下，所述主控板将所述中继板的用户接入其他的数字中继板。
  5. 根据权利要求3所述的VOIP资源的处理方法，所述方法还包括：
    所述主控板的IP交换网根据所述IP语音包中携带的端口和IP信息将所述IP语音包转发给上联板，或者，所述主控板的IP交换网根据所述IP语音包携带的端口和IP信息将所述IP语音包转发给所述主控板。
  6. 一种网络电话VOIP资源的处理装置，包括：
    转换模块，设置为获取用户语音，将所述用户语音从模拟信号转换为数字信号；
    生成模块，设置为对所述数字信号对应的语音数据流进行压缩，生成IP语音包；
    第一转发模块，设置为向主控板的IP交换网转发所述IP语音包。
  7. 根据权利要求6所述的VOIP资源的处理装置，所述装置还包括：配置模块；
    配置模块，设置为进行以下信息的配置：配置所述数字中继板的媒体IP、端口及集线比，其中，所述集线比为所述数字中继板上的中继线数与所述数字中继板上用户线数的比值，所述媒体IP和所述端口用于对所述数字中继板进行识别。
  8. 一种网络电话VOIP资源的处理装置，包括：
    记录模块，设置为记录数字中继板的集线比，并配置所述数字中继板的媒体IP和端口，其中，所述数字中继板板用于将用户语音从模拟信号转换为数字信号，并对所述数字信号对应的语音数据流进行压缩，生成IP语音包，向主控板的IP交换网转发所述IP语音包；
    接收模块，设置为接收用户的接入请求，并根据所述媒体IP、所述端口、所述集线比以及与所述用户对应的预设配置策略，配置所述用户接入的 数字中继板。
  9. 根据权利要求8所述的VOIP资源的处理装置，其中，所述预设配置策略包括以下一种或多种：
    在所述用户为预设优先处理用户的情况下，所述主控板优先给所述用户配置接入的数字中继板；
    在所述用户为数字中继板上的用户的情况下，所述主控板优先在所述数字中继板上接入所述用户；
    在所述数字中继板上存在空闲线路的情况下，所述主控板选取不是所述中继板的用户接入所述数字中继板；以及，
    在所述数字中继板上为用户配置的线路被全部占用的情况下，所述主控板将所述中继板的用户接入其他的数字中继板。
  10. 根据权利要求8所述的VOIP资源的处理装置，所述装置还包括：
    第二转发模块，设置为通过所述主控板的IP交换网根据所述IP语音包中携带的端口和IP信息将所述IP语音包转发给上联板，或者，通过所述主控板的IP交换网根据所述IP语音包携带的端口和IP信息将所述IP语音包转发给所述主控板。
  11. 一种网络电话VOIP资源处理设备，包括：多个数字中继板，主控板，其中，所述数据中继板上包括：处理器和VOIP模块；
    所述VOIP模块，设置为接收数字信号，所述数字信号是由用户语音的模拟信号转换成的；
    所述VOIP模块，还设置为对所述数字信号对应的语音数据流进行压缩，生成IP语音包；
    所述处理器，设置为将所述IP语音包转发至主控板的IP交换网；
    所述主控板，设置为记录数字中继板的集线比，并配置所述数字中继板的媒体IP和端口；其中，所述主控板处于网元中；
    所述主控板，还设置为接收用户的接入请求，并根据所述媒体IP、所述 端口、所述集线比以及与所述用户对应的预设配置策略，配置所述用户接入的数字中继板。
  12. 一种计算机可读存储介质，存储有计算机可执行指令，所述计算机可执行指令被处理器执行时实现如权利要求1至2任意一项所述的VOIP资源的处理方法以及如权利要求3至5任意一项所述的VOIP资源的处理方法。
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