CN101312428A - Voice packet transmitting and receiving method, media gateway and mobile switching center server - Google Patents
Voice packet transmitting and receiving method, media gateway and mobile switching center server Download PDFInfo
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Abstract
Description
技术领域 technical field
本发明涉及语音包的发送和接收技术,尤其涉及在具有媒体网关的软交换网络中实现语音包发送和接收的技术,属于通信技术应用领域。The invention relates to the technology of sending and receiving voice packets, in particular to the technology of realizing the sending and receiving of voice packets in a soft switch network with a media gateway, and belongs to the application field of communication technology.
背景技术 Background technique
软交换技术是网络演进以及下一代分组网络的核心技术之一,它通过软件方式实现原来交换机的控制、接续和业务处理等功能,各实体之间通过标准的协议进行连接和通信,便于在下一代网络(Next GenerationNetwork,简称NGN)中更快地实现各类复杂的协议及更方便地提供业务。Softswitch technology is one of the core technologies of network evolution and next-generation packet network. It realizes the functions of control, connection and business processing of the original switch through software, and connects and communicates with each other through standard protocols. In the next generation network (Next Generation Network, referred to as NGN), various complex protocols can be realized more quickly and services can be provided more conveniently.
软交换系统的媒体网关(Media Gateway,以下简称:MGW)是一个连接不同类型网络的单元,主要用于进行媒体转换及帧协议转换。MGW在进行语音处理时可以同时支持多种语音编解码格式,如,G.711、G.729等。The Media Gateway (hereinafter referred to as: MGW) of the softswitch system is a unit that connects different types of networks, and is mainly used for media conversion and frame protocol conversion. When performing voice processing, the MGW can simultaneously support multiple voice codec formats, such as G.711 and G.729.
例如,以G.729编解码为例,在进行语音处理时,对时分复用(TimeDivision Multiplex,简称:TDM)侧的脉冲编码调制(Pulse CodeModulation,简称:PCM)码流,采用数字信号处理器(Digital SignalProcessor,简称:DSP)来实现语音编码的转换,按固定10ms打包时长分帧,做G.729编解码转换,即遵从G.729标准协议,同时可以按配置参数做语音活动检测功能、静音压缩功能等处理,然后按照10ms,20ms或30ms多帧复用后,封装实时传输协议(Real Time Transport Protocol,简称RTP)包头及用户数据报协议(User Datagram Protocol简称:UDP)包头,然后发送给IP网络。所采用的语音包结构如表1所示,G.729算法的速率是8Kbit/s,因此每个语音包具有(20ms×8Kbit/s)/8=20字节。RTP包头固定为12字节,UDP包头固定为8字节,IP包头固定为20字节。因此,每个语音包封装成IP数据包之后总长度为60字节。For example, taking the G.729 codec as an example, when performing voice processing, a digital signal processor is used for the Pulse Code Modulation (PCM) code stream on the Time Division Multiplex (TDM) side. (Digital SignalProcessor, referred to as: DSP) to realize the conversion of voice coding, according to the fixed 10ms packaging time and frame division, do G.729 codec conversion, that is, comply with the G.729 standard protocol, and can perform voice activity detection function according to configuration parameters, Silent compression function and other processing, and then multiplex according to 10ms, 20ms or 30ms multi-frame, encapsulate Real Time Transport Protocol (RTP) header and User Datagram Protocol (User Datagram Protocol: UDP) header, and then send to the IP network. The voice packet structure adopted is shown in Table 1. The rate of the G.729 algorithm is 8Kbit/s, so each voice packet has (20ms×8Kbit/s)/8=20 bytes. The RTP header is fixed at 12 bytes, the UDP header is fixed at 8 bytes, and the IP header is fixed at 20 bytes. Therefore, after each voice packet is encapsulated into an IP data packet, the total length is 60 bytes.
表1Table 1
20byte 8byte 12byte 20字节20byte 8byte 12byte 20byte
由于这个IP数据包需要在20ms内发出,所以不考虑2层链路开销,按G.729算法计算,则每路电话带宽为60byte×8/20ms=24kbit/s。Since this IP data packet needs to be sent out within 20ms, the link overhead of Layer 2 is not considered, and calculated according to the G.729 algorithm, the bandwidth of each telephone is 60byte×8/20ms=24kbit/s.
现有技术的缺陷在于:现有语音包结构在链路上传输过程中需要增加RTP包头、UDP包头、IP包头、多协议标签交换(Multiprotocol Label Switch,MPLS)协议头、(Point to Point Protocol,简称:PPP)协议头、以太网(Ethernet)、同步数字体系(Synchronous Digital Hierarchy,简称SDH)光网络等开销,由于语音帧所占比重很小,因此实际带宽利用率很低,从而造成传输资源的巨大浪费。The defect of prior art is: existing voice packet structure needs to increase RTP packet header, UDP packet header, IP packet header, Multiprotocol Label Switching (Multiprotocol Label Switch, MPLS) protocol header, (Point to Point Protocol, Abbreviation: PPP) protocol header, Ethernet (Ethernet), Synchronous Digital Hierarchy (Synchronous Digital Hierarchy, referred to as SDH) optical network and other overheads, due to the small proportion of voice frames, the actual bandwidth utilization is very low, resulting in transmission resources huge waste.
具体地,在以太网链路上传输时,现有语音包需要增加以太网链路层开销。例如,一个G.729语音帧加上以太网数据链层开销后的总长度为78字节。加上物理层开销为98字节。因此,现有语音包净荷在以太网上的物理层利用率仅为:20/98=20.4%。在SDH光网络链路上传输时,若干个77字节的数据包填入SDH的净荷区,一个SDH帧的净荷区实际只能填充30个数据包,即77×30=2310字节,故软交换语音包净荷在155M链路物理层利用率约为:20×30/(270×9)=24.7%。Specifically, when transmitted on an Ethernet link, existing voice packets need to increase Ethernet link layer overhead. For example, the total length of a G.729 voice frame plus Ethernet data link layer overhead is 78 bytes. Plus the physical layer overhead is 98 bytes. Therefore, the physical layer utilization rate of the existing voice packet payload on the Ethernet is only: 20/98=20.4%. When transmitting on the SDH optical network link, several 77-byte data packets are filled into the SDH payload area, and the payload area of an SDH frame can actually only be filled with 30 data packets, that is, 77×30=2310 bytes Therefore, the utilization rate of the softswitch voice packet payload in the 155M link physical layer is about: 20*30/(270*9)=24.7%.
另外,由于语音通讯是一种实时业务,对时延有严格要求。按照现有标准规定,软交换侧的时延不能超过25ms,否则将影响话音质量。采用G.729格式进行的编解码的语音帧的帧长为20ms,即时延已经为20ms。因此无法通过延长帧长的方法来提高带宽利用率,因为帧长越长,时延越长。In addition, since voice communication is a real-time service, there are strict requirements on delay. According to existing standards, the time delay on the softswitch side cannot exceed 25ms, otherwise the voice quality will be affected. The frame length of the speech frame encoded and decoded in the G.729 format is 20ms, that is, the delay is already 20ms. Therefore, the bandwidth utilization cannot be improved by extending the frame length, because the longer the frame length, the longer the delay.
发明内容 Contents of the invention
本发明要解决的问题是:在不增加时延的条件下,提高软交换网络的带宽利用率。The problem to be solved by the invention is to improve the bandwidth utilization ratio of the softswitch network without increasing the time delay.
为了解决上述问题,本发明的一个实施例提供了一种语音包发送方法,包括:发端媒体网关将在同一时刻接收到的发往同一收端媒体网关的不同用户的普通语音包合成为合成语音包;将合成语音包发送给所述收端媒体网关。In order to solve the above problems, an embodiment of the present invention provides a method for sending voice packets, including: the originating media gateway synthesizes the ordinary voice packets received at the same time and sent to different users of the same receiving media gateway into a synthesized voice package; sending the synthesized voice package to the receiving media gateway.
为了解决上述问题,本发明的一个实施例提供了一种语音包接收方法,包括:收端媒体网关接受到合成语音包后,将合成语音包拆分成普通语音包;将拆分后的普通语音包进行处理或转发。In order to solve the above problems, an embodiment of the present invention provides a method for receiving voice packets, including: after receiving the synthesized voice packets, the media gateway at the receiving end splits the synthesized voice packets into ordinary voice packets; Voice packets are processed or forwarded.
为了解决上述问题,本发明的一个实施例提供了一种发端媒体网关,包括:第一通信模块,用于接收来自于不同用户的普通语音包;合成模块,用于将第一通信模块在同一时刻接收到的发往同一收端媒体网关的不同用户的普通语音包合成为合成语音包;第二通信模块,用于将由所述合成模块生成的合成语音包发送给所述收端媒体网关。In order to solve the above problems, an embodiment of the present invention provides an originating media gateway, including: a first communication module, used to receive common voice packets from different users; Synthesize voice packets received from different users sent to the same receiving media gateway at any time; the second communication module is used to send the synthesized voice packets generated by the synthesizing module to the receiving media gateway.
为了解决上述问题,本发明的一个实施例提供了一种收端媒体网关,包括:拆分模块,用于将接受到的合成语音包拆分成普通语音包;处理模块,用于将拆分模块拆分后的普通语音包进行处理或转发。In order to solve the above problems, an embodiment of the present invention provides a media gateway at the receiving end, including: a splitting module, which is used to split the received synthetic voice packets into ordinary voice packets; a processing module, which is used to split The ordinary voice packets after module splitting are processed or forwarded.
为了解决上述问题,本发明的一个实施例提供了一种媒体网关的网关能力表的生成方法,包括:移动交换中心服务器接收到的来自于媒体网关的网关能力信息后,将该网关能力信息与该媒体网关的IP地址相关联保存,生成本移动交换中心服务器上的网关能力表;当移动交换中心服务器检测到所有与其连接的媒体网关均完成网关能力信息发送后,将本动交换中心服务器上的网关能力表的所有表项信息发送给所述媒体网关;所述媒体网关将接收到的表项信息进行保存,生成本媒体网关上的网关能力表。In order to solve the above problems, an embodiment of the present invention provides a method for generating a gateway capability table of a media gateway, including: after the mobile switching center server receives the gateway capability information from the media gateway, the gateway capability information is combined with the The IP address of this media gateway is associated and preserved, and generates the gateway capability table on this mobile switching center server; Send all entry information in the gateway capability table of the media gateway to the media gateway; the media gateway saves the received entry information to generate a gateway capability table on the media gateway.
为了解决上述问题,本发明的一个实施例提供了一种媒体网关的网关能力表的更新方法,包括:媒体网关检测到本媒体网关上的网关能力表被修改后,向与其相连的移动交换中心服务器发送携带有被修改的表项信息的能力表更新请求消息;移动交换中心服务器根据能力表更新请求消息中携带的表项信息对本移动交换中心服务器上的网关能力表进行更新,并向与本移动交换中心服务器连接的除发送方媒体网关以外的其他媒体网关发送携带有更新后的表项信息的能力表更新命令消息;媒体网关根据能力表更新命令消息中携带的表项信息对本媒体网关上的网关能力表进行更新。In order to solve the above problems, an embodiment of the present invention provides a method for updating the gateway capability table of a media gateway, including: after the media gateway detects that the gateway capability table on the media gateway has been modified, it sends a report to the mobile switching center connected to it The server sends a capability table update request message carrying the modified entry information; the mobile switching center server updates the gateway capability table on the mobile switching center server according to the entry information carried in the capability table update request message, and sends a report to the mobile switching center server. Other media gateways connected to the mobile switching center server except the media gateway of the sender send a capability table update command message carrying the updated table item information; the media gateway updates the information on the media gateway according to the table item information carried in the capability table update command message. The gateway capability table of the
为了解决上述问题,本发明的一个实施例提供了一种移动交换中心服务器,包括:能力表生成模块,用于将接收到的来自于媒体网关的网关能力信息与该媒体网关的IP地址相关联保存,生成该移动交换中心服务器上的网关能力表;第一表项发送模块,用于当检测到所有与该移动交换中心服务器连接的媒体网关均完成网关能力信息发送后,将能力表生成模块生成的网关能力表的所有表项信息发送给所述媒体网关。In order to solve the above problems, an embodiment of the present invention provides a mobile switching center server, including: a capability table generation module, used to associate the received gateway capability information from the media gateway with the IP address of the media gateway Save and generate the gateway capability table on the mobile switching center server; the first table item sending module is used to send the capability table generation module when detecting that all media gateways connected to the mobile switching center server have completed the gateway capability information sending All the entry information of the generated gateway capability table is sent to the media gateway.
为了解决上述问题,本发明的一个实施例提供了另一种移动交换中心服务器,包括:能力表更新模块,用于根据来自于媒体网关的能力表更新请求消息中携带的表项信息,对该移动交换中心服务器上的网关能力表进行更新;更新命令发送模块,用于向与该移动交换中心服务器连接的除发送方媒体网关以外的其他媒体网关发送能力表更新命令消息,使媒体网关对本媒体网关上的网关能力表进行更新,其中,所述能力表更新命令消息中携带有能力表更新模块中更新后的表项信息。In order to solve the above problems, an embodiment of the present invention provides another mobile switching center server, including: a capability table update module, configured to update the The gateway capability table on the mobile switching center server is updated; the update command sending module is used to send a capability table update command message to other media gateways other than the media gateway of the sender connected to the mobile switching center server, so that the media gateway can update the media gateway for this media gateway. The gateway capability table on the gateway is updated, wherein the capability table update command message carries the updated entry information in the capability table update module.
通过本发明,由于将多个普通语音包封装为一个合成语音包在发端媒体网关和收端媒体网关之间进行传输,多个普通语音包共同使用一个IP包头,因此大大提高了实际带宽利用率,降低了传输成本。并且,通过将发送给同一收端媒体网关的不同用户的普通语音包进行合并,不需要延长帧长,也不需要等待时间。Through the present invention, since a plurality of ordinary voice packets are encapsulated into a synthesized voice packet for transmission between the media gateway at the originating end and the media gateway at the receiving end, a plurality of ordinary voice packets share one IP header, thus greatly improving the utilization rate of actual bandwidth , reducing the transmission cost. Moreover, by merging common voice packets sent to different users of the same media gateway at the receiving end, there is no need to extend the frame length or wait time.
下面通过附图和实施例,对本发明的技术方案做进一步的详细描述。The technical solutions of the present invention will be described in further detail below with reference to the accompanying drawings and embodiments.
附图说明 Description of drawings
图1为本发明实施例1所述的语音包发送及接收方法的流程图;Fig. 1 is the flow chart of voice packet sending and receiving method described in embodiment 1 of the present invention;
图2为本发明实施例2所述的网关能力表的生成方法流程图;FIG. 2 is a flowchart of a method for generating a gateway capability table according to Embodiment 2 of the present invention;
图3为本发明实施例3所述的网关能力表的更新方法流程图;FIG. 3 is a flowchart of a method for updating a gateway capability table according to Embodiment 3 of the present invention;
图4为本发明实施例1所述生成并发送合成语音包的流程图;Fig. 4 is the flow chart of generating and sending synthetic voice packets described in Embodiment 1 of the present invention;
图5为本发明实施例1所述拆分合成语音包的流程图;Fig. 5 is the flowchart of splitting and synthesizing speech packets described in Embodiment 1 of the present invention;
图6为本发明实施例4所述的发端媒体网关结构示意图;FIG. 6 is a schematic structural diagram of an originating media gateway according to Embodiment 4 of the present invention;
图7为本发明实施例5所述的收端媒体网关的结构示意图;FIG. 7 is a schematic structural diagram of the receiving media gateway according to
图8为本发明实施例6所述的MSC服务器的结构示意图;FIG. 8 is a schematic structural diagram of the MSC server described in Embodiment 6 of the present invention;
图9为本发明实施例7所述的另一种MSC服务器的结构示意图。FIG. 9 is a schematic structural diagram of another MSC server according to
具体实施方式 Detailed ways
实施例1Example 1
本实施例提供了一种语音包发送和接收方法,如图1所示,包括:The present embodiment provides a voice packet sending and receiving method, as shown in Figure 1, including:
步骤101,发端媒体网关接收到来自于用户的普通语音包后,判断该普通语音包要发往的收端媒体网关是否支持合成语音包功能,是则执行步骤110,否则执行步骤120,将接收到的普通语音包直接发送给收端媒体网关。其中,普通语音包是指采用现有帧格式的未经合并的语音包。此处需要特别指出的是,如果预先能够得知该软交换网络中的所有媒体网关均具有支持合成语音包的能力,则本步骤是可以省略的。
具体的判断方法可以有多种,例如,可以在媒体网关中建立一个网关能力表,如表2所示,There are many specific judgment methods. For example, a gateway capability table can be established in the media gateway, as shown in Table 2.
表2Table 2
在网关能力表中相关联保存有软交换网络中各个媒体网关的IP地址及是否支持合成语音包能力的网关能力信息。发端媒体网关通过对网关能力表进行查询便可以判断出相应收端媒体网关的网关能力。其中,该网关能力表可以通过手动或自动实现表项的配置。The IP address of each media gateway in the softswitch network and the gateway capability information of whether it supports the ability to synthesize voice packets are stored in the gateway capability table. The sending media gateway can determine the gateway capability of the corresponding receiving media gateway by querying the gateway capability table. Wherein, the gateway capability table can be configured manually or automatically.
步骤110,发端媒体网关将在同一时刻接收到的发往同一收端媒体网关的不同用户的普通语音包进行合并,然后发送给该收端媒体网关。具体地,可以采用如图4所示的步骤实现。
步骤110A,预先在发端媒体网关中根据本网关中保存的网关能力表中的表项信息,为每个表项中的MGW设置一个缓存队列。
步骤110B,发端媒体网关将在同一时刻接收到的发往同一收端媒体网关的普通语音包加入到相应的缓存队列中。In
步骤110C,将一个缓存队列中的普通语音包封装为一个IP数据包格式的合成语音包。其中,IP数据包的IP头中包含有收端媒体网关的IP地址。合成语音包的IP数据包格式如表3所示。
表3table 3
20字节 8字节 12字节 20字节 8字节 12字节 20字节20 bytes 8 bytes 12 bytes 20 bytes 8 bytes 12 bytes 20 bytes
如表3所示,每个普通语音包中的语音帧长度通常为20字节,但实际应用中,发往同一MGW的不同用户的语音帧可能会采用不同的编解码方式,因此,语音帧的长度也可能不同,但这并不影响对普通语音包合成。As shown in Table 3, the length of the speech frame in each common speech packet is usually 20 bytes, but in practical applications, the speech frames sent to different users of the same MGW may use different codec methods, therefore, the speech frame The length of may also be different, but this does not affect the synthesis of ordinary speech packets.
在实际的以太网应用中,上述合成语音包的最大允许长度最好不超过以太网的最大传输单元(Maximum Transmission Unit,简称:MTU)的长度。这是因为,MTU是指以太网络上允许传送的最大数据包,超过MTU长度的IP数据包就会被拆开来传送,这样会产生很多数据包分片,收端需要等待所有分片到齐后进行重新组合还原成完整的IP数据包,这无疑会增加网络的开销和时延。当MTU的长度为1500字节时,为了避免IP数据包分片重组带来的系统开销,合成语音包的长度不宜超过1500字节。假设被合成为合成语音包的普通语音包有N个,UDP头+RTP头+语音帧长=40字节,IP头为20字节,以太网二层开销为18字节,则N应该满足不等式:40×N+20<=1500-18,计算结果为:N<=36,即为了避免IP包分片重组的问题,可以被合成为合成语音包的普通语音包的个数至多为36个。合成语音包的IP净荷长度为1440字节,可填充G.729编解码20ms帧长的合成语音包36个。如果封装其他编解码的语音帧,IP净荷长度及UDP语音包个数会发生变化,但是IP净荷长度不能超过1482字节。In actual Ethernet applications, the maximum allowable length of the above-mentioned synthesized voice packet is preferably not more than the length of the Maximum Transmission Unit (MTU for short) of Ethernet. This is because MTU refers to the largest data packet that is allowed to be transmitted on the Ethernet network. IP data packets that exceed the MTU length will be disassembled and transmitted, which will generate many data packet fragments, and the receiving end needs to wait for all the fragments to arrive. Afterwards, it is reassembled and restored to a complete IP data packet, which will undoubtedly increase the overhead and delay of the network. When the length of the MTU is 1500 bytes, in order to avoid the system overhead caused by fragmentation and reassembly of IP data packets, the length of the synthesized voice packet should not exceed 1500 bytes. Assuming that there are N ordinary voice packets synthesized into synthetic voice packets, UDP header + RTP header + voice frame length = 40 bytes, IP header is 20 bytes, and Ethernet Layer 2 overhead is 18 bytes, then N should satisfy Inequality: 40×N+20<=1500-18, the calculation result is: N<=36, that is, in order to avoid the problem of fragmentation and reassembly of IP packets, the number of ordinary voice packets that can be synthesized into synthetic voice packets is at most 36 indivual. The length of the IP payload of the synthesized voice packet is 1440 bytes, which can be filled with 36 synthesized voice packets encoded and decoded by G.729 with a frame length of 20 ms. If voice frames of other codecs are encapsulated, the IP payload length and the number of UDP voice packets will change, but the IP payload length cannot exceed 1482 bytes.
步骤110D,将合成语音包发送给收端媒体网关,并清空缓存队列。
由于合成到合成语音包中的多个普通语音包是在同一时刻发送到发端媒体网关的,因此不需要等待时间,也不会增加等待时延。另外,在发端媒体网关中还可以设置一队列计时器,在预定的定时时间内,如果封装后的合成语音包的总长度达到了以太网的最大传输单元的长度,则不等定时时间到达,便开始对缓冲队列中的普通语音包进行封装。Since the multiple ordinary voice packets synthesized into the synthesized voice packet are sent to the originating media gateway at the same time, no waiting time is required, and the waiting delay will not be increased. In addition, a queue timer can also be set in the media gateway at the originating end. If the total length of the synthesized voice packet after encapsulation reaches the length of the maximum transmission unit of the Ethernet within a predetermined timing time, then the timing time will not wait until the timing arrives. It starts to encapsulate the ordinary voice packets in the buffer queue.
如果在预定的定时时间内,封装后的所述合成语音包的总长度未达到以太网的最大传输单元的长度,则当所述定时时间到达时,将所述缓存队列中的普通语音包封装为一个IP数据包格式的合成语音包。If within the predetermined timing time, the total length of the encapsulated synthetic voice packet does not reach the length of the maximum transmission unit of Ethernet, then when the timing time arrives, the ordinary voice packet in the cache queue is encapsulated A synthetic voice packet in IP packet format.
这样做的目的是为了在允许的延时条件下,合成尽可能多的普通语音包,以尽量提高网络带宽利用率。但是,为了避免延时时间过长,上述队列计时器的定时时间可以根据需要进行配置,例如可以缺省设置为5ms。设置该队列计时器的目的是为了防止在一段时间内发往某个MGW的普通语音包的数量太少导致等待时间过长,从而带来延时过长的问题。The purpose of doing this is to synthesize as many ordinary voice packets as possible under the allowable delay condition, so as to improve the utilization rate of network bandwidth as much as possible. However, in order to avoid too long delay time, the timing time of the above queue timer can be configured as required, for example, it can be set to 5ms by default. The purpose of setting the queue timer is to prevent too few ordinary voice packets sent to a certain MGW within a certain period of time, resulting in too long waiting time, thereby causing the problem of too long delay.
步骤111,收端媒体网关接受到合成语音包后,拆分成普通语音包,然后按普通语音包处理。具体地,如图5所示,可以包括如下步骤:Step 111: After receiving the synthesized voice packet, the media gateway at the receiving end splits it into ordinary voice packets, and then processes them as ordinary voice packets. Specifically, as shown in Figure 5, the following steps may be included:
步骤111A,收端媒体网关接受到来自于发端媒体网关的语音包后,判断该语音包是否为合成语音包,是则执行步骤111B,否则将该语音包按普通语音包处理。In
具体的判断方法可以是:对接收到语音包的IP包头中的总长度字段值与该语音包中的一个UDP包头中的长度字段值进行比较,如果两者之差大于IP包头的长度,例如,假设采用的IP包头的长度为20字节,上述两者之差大于20字节,则表明该语音包中至少包含有一个以上UDP数据包,说明该语音包是合成语音包;否则,如果上述两者之差等于IP包头的长度,则表明该语音包是未经合成的普通语音包。Concrete judging method can be: the total length field value in the IP packet header of receiving voice packet is compared with the length field value in a UDP packet header in this voice packet, if the difference between the two is greater than the length of IP packet header, for example , assuming that the length of the IP header used is 20 bytes, and the difference between the above two is greater than 20 bytes, it indicates that the voice packet contains at least one UDP data packet, indicating that the voice packet is a synthetic voice packet; otherwise, if If the difference between the above two is equal to the length of the IP packet header, it indicates that the voice packet is an unsynthesized ordinary voice packet.
步骤111B,将合成语音包拆分成普通语音包,按普通语音包处理。
具体地,可以通过在收端媒体网关中增加一个拆包功能层,该功能位于IP层与UDP层之间,收端媒体网关判断出接收到的语音包是合成语音包后,不将该语音包转发给UDP协议模块处理而是转发给拆包功能层处理。Specifically, by adding an unpacking function layer in the media gateway at the receiving end, the function is located between the IP layer and the UDP layer. After the media gateway at the receiving end judges that the received voice packet is a synthesized voice packet, it will not The packet is forwarded to the UDP protocol module for processing but forwarded to the unpacking functional layer for processing.
具体地,拆包功能层的功能可以通过如下算法实现:Specifically, the function of the unpacking functional layer can be realized by the following algorithm:
设L1为IP包头的长度,L2为合成语音IP包的总长度,L_ip=L2-L1,L_udp=第一个UDP包头中的长度字段中的值。Let L1 be the length of the IP packet header, L2 be the total length of the synthetic voice IP packet, L_ip=L2-L1, L_udp=the value in the length field in the first UDP packet header.
int i,j;int i, j;
i=L1,j=L1+L_udp; //设置合成语音包中的拆分点i=L1, j=L1+L_udp; //Set the split point in the synthesized voice package
L=L_ip-L_udp; //IP包头的长度与合成语音包的总长度之差L=L_ip-L_udp; //The difference between the length of the IP header and the total length of the synthetic voice packet
while(L>=0) //设置拆分条件while(L>=0) //Set split condition
{{
将i,j之间的内容发给UDP协议处理模块; //拆分出UDP语音包Send the content between i and j to the UDP protocol processing module; // split the UDP voice packet
if(L=0)break; //判断是否拆分完毕if(L=0)break; //Judge whether the split is complete
L_udp=下一个UDP包头中的长度字段中的值;L_udp=the value in the length field in the next UDP packet header;
i=j;i=j;
j=i+L_udp;//更新拆分点j=i+L_udp;//update split point
L=L-L_udp;L=L-L_udp;
}}
通过本实施例所述方法,由于将多个普通语音包封装为一个合成语音包在发端媒体网关和收端媒体网关之间进行传输,多个普通语音包共同使用一个I P包头,因此大大提高了实际带宽利用率。分析表明,采用本实施所述的合成语音包在以太网链路进行传输时,所需占用的带宽仅为现有方法的0.42;在SDH链路上进行传输时,所需占用的带宽仅为现有方法的0.54,即节省约一半左右的带宽资源,从而大大提高了实际的带宽利用率,降低了传输成本。By the method described in the present embodiment, because a plurality of ordinary voice packets are encapsulated into a synthetic voice packet for transmission between the media gateway of the originating end and the media gateway of the receiving end, a plurality of ordinary voice packets share an IP packet header, thus greatly improving actual bandwidth utilization. Analysis shows that when the synthesized voice packet described in this implementation is used to transmit on the Ethernet link, the required bandwidth is only 0.42 of the existing method; when transmitted on the SDH link, the required bandwidth is only 0.54 of the existing method, which saves about half of the bandwidth resources, thus greatly improving the actual bandwidth utilization and reducing the transmission cost.
并且,本实施例所述方法是将发送给同一收端媒体网关的不同用户的普通语音包进行了合并,在实际情况中,由于电信级MGW的话务量非常大,发端媒体网关在同一时刻接收到的来自于不同用户的普通语音包的数量非常多,将这些普通语音包进行合并,除了需要一定的合成时间外,几乎不需要延长帧长,也不需要等待时间。而其中的合成时间仅为微秒量级,与毫秒量级的语音帧相比几乎可以忽略。Moreover, the method described in this embodiment is to combine the ordinary voice packets sent to different users of the same receiving media gateway. There are a lot of common speech packets received from different users. Combining these common speech packets requires almost no frame length extension or waiting time, except for a certain synthesis time. The synthesis time is only on the order of microseconds, which is almost negligible compared with the speech frames on the order of milliseconds.
实施例2Example 2
上述步骤101中所述的网关能力表的具体的生成和更新方法,可以由媒体网关通过与移动交换中心(Mobile Switching Center,简称:MSC)服务器相配合实现。其中,MSC服务器是MSC中的一个功能实体,它与MGW共同实现软交换核心网中的交换功能,以下进行具体说明:The specific method for generating and updating the gateway capability table described in the
网关能力表的生成过程如图2所示,The generation process of the gateway capability table is shown in Figure 2.
步骤201,MGW向与其连接的MSC服务器发送本网关的网关能力信息,用于通知MSC服务器本网关是否具有支持合成语音包的能力。In
具体地,可以通过现有协议中的服务器业务变更请求(ServerServiceChange.Req)命令中的非标准数据(Non Standard Data)参数实现。例如,可以将非标准数据参数设置为“MGW名称,MGW IP地址,支持合成语音包”,即表明具有该IP地址的MGW支持合成语音包功能。Specifically, it can be realized through the non-standard data (Non Standard Data) parameter in the server service change request (ServerServiceChange.Req) command in the existing protocol. For example, the non-standard data parameter can be set as "MGW name, MGW IP address, support synthesized voice package", which means that the MGW with this IP address supports the function of synthesized voice package.
步骤202,MSC服务器将接收到的来自于MGW的网关能力信息与该MGW的IP地址相关联保存在本服务器上的网关能力表中,即生成如表4所示的网关能力表。In
步骤203,当MSC服务器检测到所有与其连接的MGW均完成网关能力信息发送后,将网关能力表的所有表项信息发送MGW进行保存。
步骤204,如果一个软交换网络中存在多个MSC服务器,则当一个MSC服务器完成网关能力表生成后,还可以将网关能力表的表项信息发送给相邻MSC服务器。
具体地,可以通过与承载无关的呼叫控制(Bearer Independent CallControl Protocol,简称:BICC)协议中的初始地址管理(Initial AddressManagement简称,IAM)消息实现。例如,可以在IAM消息的被叫号码(CalledNumber,简称CLD)字段填写收端MSC服务器的局号,编解码列表(CodecList)字段填写“MGW名称,MGW IP地址,支持合成语音包”。Specifically, it may be implemented through an initial address management (Initial Address Management, IAM) message in a Bearer Independent Call Control Protocol (BICC for short) protocol. For example, the station number of the receiving MSC server can be filled in the called number (CalledNumber, CLD for short) field of the IAM message, and the codec list (CodecList) field can be filled with "MGW name, MGW IP address, and support for synthetic voice packets".
步骤205,相邻MSC服务器将接收到的表项信息保存到本服务器上,然后通过与步骤203类似的方式通知与其相连的MGW,并由将MGW将网关能力表从该相邻MSC服务器复制到本网关进行保存。
此处需要特别指出的是,已经收到IAM消息的MSC服务器不再向其他相邻的MSC服务器发送IAM消息,这样做是为避免系统中出现环路或死循环。如果收端MSC服务器需要向发端MSC服务器发送本域内的MGW网关能力信息集,可以通过现有协议中的应用传送机制(Application TransportMechanism,简称:APM)消息实现。What needs to be pointed out here is that the MSC server that has received the IAM message will not send IAM messages to other adjacent MSC servers. This is to avoid loops or infinite loops in the system. If the receiving MSC server needs to send the MGW gateway capability information set in the local domain to the sending MSC server, it can be realized through the Application Transport Mechanism (Application Transport Mechanism, APM for short) message in the existing protocol.
此处需要特别指出的是,通过步骤201-205实现了在MGW上配置网关能力表,但不通过上述方法而是通过手动将网关能力表配置到各个MGW上也是可以的,而通过本实施例所述方法可以自动实现网关能力表的配置,有利于提高配置效率。What needs to be pointed out here is that the gateway capability table is configured on the MGW through steps 201-205, but it is also possible to configure the gateway capability table on each MGW manually instead of the above method, and through this embodiment The method can automatically realize the configuration of the gateway capability table, which is beneficial to improve configuration efficiency.
通过本实施例所述方法,实现了自动生成网关能力表,与手动配置相比,速度更快,效率更高,准确性更高,并且节省人力成本。Through the method described in this embodiment, the gateway capability table is automatically generated, which is faster, more efficient, and more accurate than manual configuration, and saves labor costs.
实施例3Example 3
本实施例提供了一种对媒体网关上的网关能力表进行更新的方法,如图3所示,包括如下步骤:This embodiment provides a method for updating the gateway capability table on the media gateway, as shown in Figure 3, including the following steps:
步骤301,MGW检测到本网关上保存的网关能力表被修改后,向MSC服务器发送能力表更新请求消息,其中携带有被修改的表项信息。具体地,MGW可以通过现有协议中的通知请求消息(Notify.reqeust)上报MSC服务器,请求更新网关能力表的相关信息。In
其中,用户对MGW上保存的网关能力表进行修改,可以通过在MGW上配置的用户界面实现,具体地,用户可以通过用户界面开启或关闭本网关的合成语音包功能。Wherein, the user can modify the gateway capability table saved on the MGW through the user interface configured on the MGW. Specifically, the user can enable or disable the synthesized voice packet function of the gateway through the user interface.
步骤302,MSC服务器接收到来自于MGW的能力表更新请求消息后,根据其中携带的表项信息对本服务器上保存的网关能力表的相应表项进行更新。In
步骤303,MSC服务器完成网关能力表更新后,向与该MSC服务器连接的其他MGW发送携带有更新后的表项信息的能力表更新命令消息。具体地,能力表更新命令消息可以采用现有协议中的修改请求消息(Modify.request)。
另外,MSC服务器还可以根据需要向请求更新表项的MGW回复能力表更新确认消息,以通知该MGW网关能力表已经完成了更新。具体地,能力表更新确认消息可以采用现有协议中的通知应答消息(Notify.reply)。In addition, the MSC server may also reply a capability table update acknowledgment message to the MGW requesting to update the entry as needed, so as to notify the MGW that the gateway capability table has been updated. Specifically, the capability table update confirmation message may adopt a notification response message (Notify.reply) in the existing protocol.
步骤304,MGW接收到能力表更新命令消息后,根据其中携带的表项信息对本网关上保存的网关能力表进行更新。
步骤305,如果一个软交换网络中存在多个MSC服务器,则一个MSC服务器完成网关能力表更新后,还可以将更新后的表项信息发送给相邻MSC服务器,由该相邻MSC服务器对本服务器及与其相连的MGW上的网关能力表进行更新。具体地,可以按照类似于步骤205所述方法,通过IAM消息实现表项信息传输。
此处需要特别指出的是,如果用户修改的是MSC服务器上保存的网关能力表而不是MGW上保存的网关能力表,则MSC服务器可以直接向与其连接所有其他MGW发送携带有更新后的表项信息的能力表更新命令消息;MGW接收到能力表更新命令消息后,根据其中携带的表项信息对本网关上保存的网关能力表进行更新。具体地,能力表更新命令消息可以采用现有协议中的修改请求消息(Modify.request)。It should be pointed out here that if the user modifies the gateway capability table saved on the MSC server instead of the gateway capability table saved on the MGW, the MSC server can directly send the updated table entries to all other MGWs connected to it. information of a capability table update command message; after receiving the capability table update command message, the MGW updates the gateway capability table stored on the gateway according to the entry information carried therein. Specifically, the capability table update command message may adopt a modification request message (Modify.request) in the existing protocol.
通过本实施例所述方法,实现了自动的网关能力表的更新,与手动配置相比,速度更快,效率更高,准确性更高,并且当网络结构发生变化时可以自动实现更新,更新更加及时,节省人力成本。Through the method described in this embodiment, the automatic update of the gateway capability table is realized. Compared with the manual configuration, the speed is faster, the efficiency is higher, and the accuracy is higher. When the network structure changes, the update can be automatically implemented. More timely and save labor costs.
实施例4Example 4
本实施例提供了一种发端媒体网关,如图6所示,发端媒体网关4包括:第一通信模块41,合成模块42和第二通信模块43。This embodiment provides an originating media gateway. As shown in FIG. 6 , the originating media gateway 4 includes: a
其中,第一通信模块41用于接收来自于不同用户的普通语音包;合成模块42,用于将第一通信模块41在同一时刻接收到的发往同一收端媒体网关的不同用户的普通语音包合成为合成语音包;第二通信模块43,用于将合成模块42生成的合成语音包发送给所述收端媒体网关。具体地,为了避免IP数据包在进行以太网传输时由于分片重组带来的系统开销,因此,当第一通信模块41中封装后的合成语音包的总长度达到了以太网的最大传输单元的长度时,第二通信模块43便可以将其进行发送。Wherein, the
具体地,合成模块42可以具体包括:缓存模块42A和封装模块42B。其中,缓存模块42A用于为收端媒体网关设置缓存队列,并将第一通信模块41在同一时刻接收到的发往同一收端媒体网关的不同用户的普通语音包加入到相应的缓存队列中;封装模块42B用于将所述缓存模块42A中的一个缓存队列中的普通语音包封装为一个IP数据包格式的合成语音包,其中,该IP数据包的IP头中包含有所述收端媒体网关的IP地址。Specifically, the
另外,合成模块42中还可以设置定时模块42C:用于设置队列计时器,当该队列计时器的定时时间到达时,向封装模块42B发送定时信号;封装模块42B收到该定时信号后,尽管将此时的缓存模块42A中的缓存队列中的普通语音包封装后的合成语音包的总长度未达到以太网的最大传输单元的长度,但仍然开始执行封装合成语音包的步骤。这样做的目的是为了在允许的延时条件下,合成尽可能多的普通语音包,以尽量提高网络带宽利用率。但是,为了避免延时时间过长,上述队列计时器的定时时间可以根据需要进行配置,例如可以缺省设置为5ms。设置该队列计时器的目的是为了防止在一段时间内发往某个MGW的普通语音包的数量太少导致等待时间过长,从而带来延时过长的问题。In addition, a
另外,该发端媒体网关4中还可以进一步包括能力判断模块44,用于判断第一通信模块41接收到的来自于用户的普通语音包要发往的收端媒体网关是否支持合成语音包功能,是则向合成模块42发送判断结果,使所述合成模块42开始执行封装合成语音包的步骤;否则将第一通信模块41接收到的普通语音包直接发送给所述收端媒体网关。In addition, the originating media gateway 4 may further include a
通过本实施例所述的发端媒体网关,实现了将普通语音包封装为合成语音包,大大提高了实际带宽利用率,并且不会产生延迟。Through the originating media gateway described in this embodiment, it is realized that ordinary voice packets are encapsulated into synthesized voice packets, which greatly improves the utilization rate of the actual bandwidth and does not cause delay.
实施例5Example 5
本实施例提供了一种收端媒体网关,如图7所示,该收端媒体网关5包括:拆分模块51和处理模块52。This embodiment provides a receiving media gateway. As shown in FIG. 7 , the receiving
其中,拆分模块51用于将接受到的合成语音包拆分成普通语音包;处理模块52,用于将拆分模块51拆分后的普通语音包进行处理或转发。Wherein, the
另外,该收端媒体网关5还可以进一步包括语音包判断模块53,用于判断接受到的来自于发端媒体网关的语音包是否为合成语音包,是则将该合成语音包发送给拆分模块51。In addition, the
通过本实施例所述的收端媒体网关,实现了将合成语音包拆分成普通语音包,完成普通语音包的接收和处理。Through the media gateway at the receiving end described in this embodiment, it is realized that the synthesized voice packets are divided into ordinary voice packets, and the receiving and processing of the ordinary voice packets is completed.
实施例6Example 6
本实施例提供了一种移动交换中心服务器,即MSC服务器,如图8所示,MSC服务器6包括:能力表生成模块61和第一表项发送模块62。This embodiment provides a mobile switching center server, that is, an MSC server. As shown in FIG. 8 , the MSC server 6 includes: a capability table generating module 61 and a first entry sending module 62 .
其中,能力表生成模块61用于将接收到的来自于MGW的网关能力信息与该MGW的IP地址相关联保存,生成本服务器上的网关能力表;第一表项发送模块62用于当检测到所有与本服务器连接的MGW均完成网关能力信息发送后,将能力表生成模块61生成的网关能力表的所有表项信息发送给各个MGW。Among them, the capability table generation module 61 is used to associate and store the received gateway capability information from the MGW with the IP address of the MGW, and generate a gateway capability table on the server; the first entry sending module 62 is used to detect After all the MGWs connected to the server have finished sending the gateway capability information, all the item information of the gateway capability table generated by the capability table generation module 61 is sent to each MGW.
另外,如果一个网络中存在多个MSC服务器,则当一个MSC服务器完成网关能力表生成后,还可以将网关能力表的表项信息发送给相邻MSC服务器。因此,本实施例所述的MSC服务器6还可以进一步包括第二表项发送模块63,用于当检测到所有与本服务器连接的MGW均完成网关能力信息发送后,将能力表生成模块61生成的网关能力表的所有表项信息发送给相邻MSC服务器。相应地,MSC服务器6中还可以设置第三表项发送模块64,用于将来自于相邻MSC服务器的表项信息发送给与本服务器相连的所有MGW,使这些MGW生成本网关上的网关能力表。In addition, if there are multiple MSC servers in a network, after one MSC server completes the generation of the gateway capability table, it can also send the entry information of the gateway capability table to the adjacent MSC server. Therefore, the MSC server 6 described in this embodiment may further include a second entry sending module 63, which is used to generate the capability table generating module 61 after detecting that all MGWs connected to the server have completed sending gateway capability information. All entry information in the gateway capability table of the gateway is sent to the adjacent MSC server. Correspondingly, a third entry sending module 64 can also be set in the MSC server 6, which is used to send the entry information from the adjacent MSC server to all MGWs connected to this server, so that these MGWs can generate gateways on this gateway. Ability table.
通过本实施例所述的MSC服务器,实现了自动生成网关能力表。适用于具有一个或多个MSC服务器的网络。Through the MSC server described in this embodiment, automatic generation of the gateway capability table is realized. Suitable for networks with one or more MSC servers.
实施例7Example 7
本实施例提供了另一种MSC服务器7,如图9所示,包括:能力表更新模块71和更新命令发送模块72。This embodiment provides another
其中,能力表更新模块71用于根据来自于MGW的能力表更新请求消息中携带的表项信息,对本服务器上的网关能力表进行更新;更新命令发送模块72用于向与本服务器连接的除发送方MGW以外的其他MGW发送能力表更新命令消息,使MGW对本网关上的网关能力表进行更新,其中,所述能力表更新命令消息中携带有能力表更新模块71中更新后的表项信息。Among them, the capability
另外,MSC服务器7还可以包括:更新信息发送模块73,用于将能力表更新模块71中更新后的表项信息发送给相邻MSC服务器,使相邻MSC服务器可以根据该表项信息对与该相邻MSC服务器相连的MGW上的网关能力表进行更新。In addition, the
通过本实施例所述的MSC服务器,实现了自动更新网关能力表。适用于具有一个或多个MSC服务器的网络。Through the MSC server described in this embodiment, automatic updating of the gateway capability table is realized. Suitable for networks with one or more MSC servers.
最后所应说明的是,以上实施例仅用以说明本发明的技术方案而非限制,尽管参照较佳实施例对本发明进行了详细说明,本领域的普通技术人员应当理解,可以对本发明的技术方案进行修改或者等同替换,而不脱离本发明技术方案的精神和范围。Finally, it should be noted that the above embodiments are only used to illustrate the technical solutions of the present invention without limitation. Although the present invention has been described in detail with reference to the preferred embodiments, those of ordinary skill in the art should understand that the technical solutions of the present invention can be The scheme shall be modified or equivalently replaced without departing from the spirit and scope of the technical scheme of the present invention.
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