CN110012180A - A kind of connection method, device, equipment and the medium of voip network phone - Google Patents

A kind of connection method, device, equipment and the medium of voip network phone Download PDF

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Publication number
CN110012180A
CN110012180A CN201910294473.XA CN201910294473A CN110012180A CN 110012180 A CN110012180 A CN 110012180A CN 201910294473 A CN201910294473 A CN 201910294473A CN 110012180 A CN110012180 A CN 110012180A
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China
Prior art keywords
called party
party
call request
sip server
current address
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杨武
青亮
周禹
古恒
杨俊�
何伟
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Chengdu Westone Information Industry Inc
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Chengdu Westone Information Industry Inc
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Priority to CN201910294473.XA priority Critical patent/CN110012180A/en
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    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04LTRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
    • H04L65/00Network arrangements, protocols or services for supporting real-time applications in data packet communication
    • H04L65/1066Session management
    • H04L65/1101Session protocols
    • H04L65/1104Session initiation protocol [SIP]
    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04MTELEPHONIC COMMUNICATION
    • H04M7/00Arrangements for interconnection between switching centres
    • H04M7/006Networks other than PSTN/ISDN providing telephone service, e.g. Voice over Internet Protocol (VoIP), including next generation networks with a packet-switched transport layer

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  • Engineering & Computer Science (AREA)
  • Computer Networks & Wireless Communication (AREA)
  • Signal Processing (AREA)
  • Business, Economics & Management (AREA)
  • General Business, Economics & Management (AREA)
  • Multimedia (AREA)
  • Telephonic Communication Services (AREA)

Abstract

This application discloses connection method, device, equipment and the computer readable storage mediums of a kind of voip network phone.It is different from the prior art, in this application, if the call request of calling party can not be sent to callee by sip server, notify the third party device with Android backstage that the call request of calling party is sent to callee, and calling party and callee are thus established voip network telephone connection.Pass through the method in the application, can occur in the NAT aging either killed situation in Android backstage in callee, calling party and callee can also be established voip network telephone connection, so, substantially increase the call completion ratio of calling party and callee's voip network phone.

Description

一种VoIP网络电话的连接方法、装置、设备及介质A connection method, device, equipment and medium of a VoIP internet phone

技术领域technical field

本公开涉及通讯技术领域,特别涉及一种VoIP网络电话的连接方法、装置、设备及介质。The present disclosure relates to the field of communication technologies, and in particular, to a connection method, device, device and medium for a VoIP network phone.

背景技术Background technique

VoIP网络电话(Voice over internet Protocol,VoIP协议)是一种将模拟信号经过压缩与封包之后,以数据封包的形式在IP网络中进行语音讯号的传输过程。参与通信的主叫方和被叫方在建立VoIP网络电话连接的过程中,被叫方由于安卓后台被杀掉或者是由于NAT(Network Address Translation,网络地址转换)端口老化等问题,使得SIP服务器(Session Initiation Protocol,会话初始协议)无法通过被叫方的注册地址找到被叫方,从而使得主叫方的呼叫请求不能够到达被叫方,由此导致主叫方和被叫方的VoIP网络电话呼通率较低。由此可见,通过怎样的一种方法来提高主叫方和被叫方的VoIP网络电话呼通率,是本领域技术人员亟待解决的问题。VoIP (Voice over internet Protocol, VoIP protocol) is a process of transmitting voice signals in the IP network in the form of data packets after the analog signals are compressed and packaged. During the process of establishing a VoIP network telephone connection between the calling party and the called party participating in the communication, the called party is killed due to the Android background or the aging of the NAT (Network Address Translation) port, which makes the SIP server (Session Initiation Protocol, Session Initiation Protocol) The called party cannot be found through the registered address of the called party, so that the call request of the calling party cannot reach the called party, thus causing the VoIP network of the calling party and the called party. The call-through rate is low. It can be seen that how to improve the call-through rate of the calling party and the called party through the VoIP network is an urgent problem to be solved by those skilled in the art.

发明内容SUMMARY OF THE INVENTION

有鉴于此,本发明的目的在于提供一种VoIP网络电话的连接方法、装置、设备及介质,以提高VoIP网络电话的呼通率。其具体方案如下:In view of this, the purpose of the present invention is to provide a connection method, device, device and medium for a VoIP network phone, so as to improve the call-through rate of the VoIP network phone. Its specific plan is as follows:

一种VoIP网络电话的连接方法,应用于SIP服务器,包括:A connection method for a VoIP internet phone, applied to a SIP server, comprising:

当接收到主叫方向被叫方发送的第一呼叫请求时,向所述被叫方的注册地址发送所述第一呼叫请求,并判断所述第一呼叫请求是否到达所述被叫方;When receiving the first call request sent by the calling party to the called party, send the first call request to the registered address of the called party, and determine whether the first call request reaches the called party;

若所述第一呼叫请求未到达所述被叫方,则通知具有安卓后台的第三方设备向所述被叫方发送目标指令,以获取所述被叫方的当前地址;If the first call request does not reach the called party, notifying a third-party device with an Android background to send a target instruction to the called party to obtain the current address of the called party;

当获取到所述被叫方的当前地址时,则向所述被叫方的当前地址发送第二呼叫请求,以在所述主叫方和所述被叫方之间建立VoIP网络电话连接。When the current address of the called party is acquired, a second call request is sent to the current address of the called party, so as to establish a VoIP network telephone connection between the calling party and the called party.

可选的,所述通知具有安卓后台的第三方设备向所述被叫方发送目标指令,以获取所述被叫方的当前地址,包括:Optionally, a third-party device with an Android background in the notification sends a target instruction to the called party to obtain the current address of the called party, including:

通知所述第三方设备向所述被叫方发送所述目标指令;Notifying the third-party device to send the target instruction to the called party;

接收所述被叫方基于所述目标指令发送的注册请求;receiving a registration request sent by the called party based on the target instruction;

在注册成功后,将所述被叫方新注册的地址确定为所述被叫方的当前地址。After the registration is successful, the newly registered address of the called party is determined as the current address of the called party.

可选的,所述通知具有安卓后台的第三方设备向所述被叫方发送目标指令,包括:Optionally, the notification that a third-party device with an Android background sends a target instruction to the called party, including:

通知具有安卓后台的所述第三方设备的推送服务向所述被叫方发送所述目标指令。Notifying the push service of the third-party device with the Android background to send the target instruction to the called party.

可选的,所述通知具有安卓后台的第三方设备向所述被叫方发送目标指令,包括:Optionally, the notification that a third-party device with an Android background sends a target instruction to the called party, including:

通知具有安卓后台的所述第三方设备的短信服务向所述被叫方发送所述目标指令。Notifying the short message service of the third-party device with the Android background to send the target instruction to the called party.

可选的,在所述通知具有安卓后台的第三方设备向所述被叫方发送目标指令之后,还包括:Optionally, after the notification that the third-party device with the Android background sends the target instruction to the called party, further includes:

判断是否在预设时长内获取到所述被叫方的当前地址;Determine whether the current address of the called party is obtained within a preset time period;

若未在所述预设时长内获取到所述当前地址,则向所述主叫方返回呼叫失败信息。If the current address is not obtained within the preset time period, call failure information is returned to the calling party.

为实现上述目的,本公开还提供了一种VoIP网络电话的连接方法,应用于被叫方,包括:To achieve the above purpose, the present disclosure also provides a method for connecting a VoIP network phone, which is applied to the called party, including:

接收独立于SIP服务器的具有安卓后台的第三方设备发送的目标指令,所述目标指令用于指示所述SIP服务器向所述被叫方发送主叫方的呼叫请求;Receive a target instruction sent by a third-party device with an Android background independent of the SIP server, where the target instruction is used to instruct the SIP server to send a call request of the calling party to the called party;

根据所述目标指令,运行VoIP应用;According to the target instruction, run the VoIP application;

向所述SIP服务器发送注册请求,注册新的当前地址,以使所述SIP服务器向所述当前地址发送所述主叫方的呼叫请求。A registration request is sent to the SIP server to register a new current address, so that the SIP server sends a call request of the calling party to the current address.

为实现上述目的,本公开还提供了一种VoIP网络电话的连接装置,应用于SIP服务器,包括:In order to achieve the above object, the present disclosure also provides a connection device for a VoIP Internet phone, which is applied to a SIP server, including:

请求发送模块,用于当接收到主叫方向被叫方发送的第一呼叫请求时,向所述被叫方的注册地址发送所述第一呼叫请求,并判断所述第一呼叫请求是否到达所述被叫方;A request sending module, configured to send the first call request to the registered address of the called party when receiving the first call request sent by the calling party to the called party, and determine whether the first call request arrives the called party;

地址获取模块,用于若所述第一呼叫请求未到达所述被叫方,则通知具有安卓后台的第三方设备向所述被叫方发送目标指令,以获取所述被叫方的当前地址;The address obtaining module is configured to notify the third-party device with an Android background to send a target instruction to the called party to obtain the current address of the called party if the first call request does not reach the called party ;

电话建立模块,用于当获取到所述被叫方的当前地址时,则向所述被叫方的当前地址发送第二呼叫请求,以在所述主叫方和所述被叫方之间建立VoIP网络电话连接。The phone establishment module is configured to send a second call request to the current address of the called party when the current address of the called party is obtained, so as to establish a connection between the calling party and the called party Establish a VoIP internet phone connection.

可选的,所述地址获取模块,包括:Optionally, the address obtaining module includes:

指令发送单元,用于通知所述第三方设备向所述被叫方发送所述目标指令;an instruction sending unit, configured to notify the third-party device to send the target instruction to the called party;

请求注册单元,用于接收所述被叫方基于所述目标指令发送的注册请求;a registration request unit, configured to receive a registration request sent by the called party based on the target instruction;

地址更新单元,用于在注册成功后,将所述被叫方新注册的地址确定为所述被叫方的当前地址。The address updating unit is configured to determine the newly registered address of the called party as the current address of the called party after the registration is successful.

可选的,所述地址获取模块,包括:Optionally, the address obtaining module includes:

第一指令发送单元,用于通知具有安卓后台的所述第三方设备的推送服务向所述被叫方发送所述目标指令。A first instruction sending unit, configured to notify the push service of the third-party device with an Android background to send the target instruction to the called party.

可选的,所述地址获取模块,包括:Optionally, the address obtaining module includes:

第二指令发送单元,用于通知具有安卓后台的所述第三方设备的短信服务向所述被叫方发送所述目标指令。The second instruction sending unit is configured to notify the short message service of the third-party device with the Android background to send the target instruction to the called party.

可选的,该VoIP网络电话的连接装置还包括:Optionally, the connection device of the VoIP Internet phone further includes:

时长判断模块,用于在所述通知具有安卓后台的第三方设备向所述被叫方发送目标指令之后,判断是否在预设时长内获取到所述被叫方的当前地址;A duration judging module, configured to determine whether the current address of the called party is obtained within a preset duration after the notification that the third-party device with an Android background sends a target instruction to the called party;

信息返回模块,用于若未在所述预设时长内获取到所述当前地址,则向所述主叫方返回呼叫失败信息。An information return module, configured to return call failure information to the calling party if the current address is not obtained within the preset time period.

为实现上述目的,本公开还提供了一种VoIP网络电话的连接装置,应用于被叫方,包括:In order to achieve the above object, the present disclosure also provides a connection device for a VoIP network phone, which is applied to the called party, including:

指令接收模块,用于接收独立于SIP服务器的具有安卓后台的第三方设备发送的目标指令,所述目标指令用于指示所述SIP服务器向所述被叫方发送主叫方的呼叫请求;an instruction receiving module, configured to receive a target instruction sent by a third-party device with an Android background independent of the SIP server, where the target instruction is used to instruct the SIP server to send a call request of the calling party to the called party;

应用运行模块,用于根据所述目标指令,运行VoIP应用;an application running module for running the VoIP application according to the target instruction;

地址注册模块,用于向所述SIP服务器发送注册请求,注册新的当前地址,以使所述SIP服务器向所述当前地址发送所述主叫方的呼叫请求。The address registration module is configured to send a registration request to the SIP server to register a new current address, so that the SIP server sends a call request of the calling party to the current address.

为实现上述目的,本公开还提供了一种VoIP网络电话的连接设备,包括:To achieve the above purpose, the present disclosure also provides a connection device for a VoIP internet phone, including:

存储器,用于存储计算机程序;memory for storing computer programs;

处理器,用于执行所述计算机程序时实现如上述内容所描述的一种VoIP网络电话的连接方法的步骤。The processor is configured to implement the steps of the method for connecting a VoIP internet phone as described in the above content when executing the computer program.

为实现上述目的,本公开还提供了一种计算机可读存储介质,所述计算机可读存储介质上存储有计算机程序,所述计算机程序被处理器执行时实现如上述内容所描述的VoIP网络电话的连接方法的步骤。In order to achieve the above object, the present disclosure also provides a computer-readable storage medium, where a computer program is stored on the computer-readable storage medium, and when the computer program is executed by a processor, the VoIP network telephone as described above is realized. the steps of the connection method.

通过上述技术方案可知,在本公开中当SIP服务器接收到主叫方向被叫方发送的第一呼叫请求时,首先是将第一呼叫请求发送至被叫方在SIP服务器上的注册地址,并判断第一呼叫请求是否到达被叫方,如果第一呼叫请求没有到达被叫方,则说明被叫方的NAT老化或者是安卓后台被杀死,此时,SIP服务器通过调用具有安卓后台的第三方设备将第一呼叫请求发送至被叫方,以使得被叫方将当前地址发送至SIP服务器。当SIP服务器获取到被叫方的当前地址时,就可以向被叫方的当前地址发送第二呼叫请求,并以此来将主叫方和被叫方建立VoIP网络电话连接。显然,相比于现有技术中,如果被叫方出现NAT老化或者是安卓后台被杀死的情况下,SIP服务器无法将呼叫请求发送至被叫方而言,在本公开中SIP服务器是通过调用具有安卓后台的第三方设备来将主叫方的呼叫请求发送至被叫方,由此大大了提高主叫方和被叫方VoIP网络电话的呼通率。As can be seen from the above technical solutions, in the present disclosure, when the SIP server receives the first call request sent by the calling party to the called party, it first sends the first call request to the registered address of the called party on the SIP server, and Determine whether the first call request reaches the called party. If the first call request does not reach the called party, it means that the NAT of the called party is aging or the Android background is killed. At this time, the SIP server calls the No. The three-party device sends the first call request to the called party, so that the called party sends the current address to the SIP server. When the SIP server obtains the current address of the called party, it can send a second call request to the current address of the called party, and thereby establish a VoIP network telephone connection between the calling party and the called party. Obviously, compared with the prior art, if the called party has NAT aging or the Android background is killed, the SIP server cannot send the call request to the called party. In the present disclosure, the SIP server uses the A third-party device with an Android background is called to send the calling party's call request to the called party, thereby greatly improving the call-through rate of the calling party and the called party's VoIP network phone.

本公开还同时提供了一种VoIP网络电话的连接装置、设备及介质,具有相同的有益效果,在此不再赘述。The present disclosure also provides a connection device, equipment and medium for a VoIP internet phone, which have the same beneficial effects, and will not be repeated here.

附图说明Description of drawings

为了更清楚地说明本发明实施例或现有技术中的技术方案,下面将对实施例或现有技术描述中所需要使用的附图作简单地介绍,显而易见地,下面描述中的附图仅仅是本发明的实施例,对于本领域普通技术人员来讲,在不付出创造性劳动的前提下,还可以根据提供的附图获得其他的附图。In order to explain the embodiments of the present invention or the technical solutions in the prior art more clearly, the following briefly introduces the accompanying drawings that need to be used in the description of the embodiments or the prior art. Obviously, the accompanying drawings in the following description are only It is an embodiment of the present invention. For those of ordinary skill in the art, other drawings can also be obtained according to the provided drawings without creative work.

图1为本发明实施例一提供的一种VoIP网络电话的连接方法的流程图;1 is a flowchart of a method for connecting a VoIP phone according to Embodiment 1 of the present invention;

图2为本发明实施例一提供的一种SIP服务器通知第三方设备呼叫被叫方的示意图;2 is a schematic diagram of a SIP server notifying a third-party device to call a called party according to Embodiment 1 of the present invention;

图3为本发明实施例一提供的一种SIP服务器重新呼叫被叫方的示意图;3 is a schematic diagram of a SIP server re-calling a called party according to Embodiment 1 of the present invention;

图4为本发明实施例二提供的一种VoIP网络电话的连接方法的流程图;4 is a flowchart of a method for connecting a VoIP phone according to Embodiment 2 of the present invention;

图5为本发明实施例三提供的一种VoIP网络电话的连接方法的流程图;5 is a flowchart of a method for connecting a VoIP phone according to Embodiment 3 of the present invention;

图6为本发明实施例四提供的一种VoIP网络电话的连接方法的流程图;6 is a flowchart of a method for connecting a VoIP phone according to Embodiment 4 of the present invention;

图7为本发明实施例五提供的一种VoIP网络电话的连接方法的流程图;7 is a flowchart of a method for connecting a VoIP phone according to Embodiment 5 of the present invention;

图8为本发明实施例六提供的一种VoIP网络电话的连接方法的流程图;8 is a flowchart of a method for connecting a VoIP internet phone according to Embodiment 6 of the present invention;

图9为本发明实施例提供的一种VoIP网络电话的连接装置的结构图;9 is a structural diagram of a connection device for a VoIP internet phone provided by an embodiment of the present invention;

图10为本发明实施例提供的另一种VoIP网络电话的连接装置的结构图;10 is a structural diagram of another VoIP internet phone connection device provided by an embodiment of the present invention;

图11为本发明实施例提供的一种VoIP网络电话的连接设备的结构图。FIG. 11 is a structural diagram of a connection device of a VoIP internet phone according to an embodiment of the present invention.

具体实施方式Detailed ways

本申请的核心是提供一种VoIP网络电话的连接方法、装置、设备及计算机可读存储介质。区别于现有技术,在本申请中,如果SIP服务器无法将主叫方的呼叫请求发送至被叫方,则通知具有安卓后台的第三方设备将主叫方的呼叫请求发送至被叫方,来将主叫方和被叫方建立VoIP网络电话连接,并由此来提高主叫方和被叫方的VoIP网络电话呼通率。The core of this application is to provide a connection method, apparatus, device and computer-readable storage medium for a VoIP internet phone. Different from the prior art, in the present application, if the SIP server cannot send the call request of the calling party to the called party, it notifies the third-party device with the Android background to send the calling request of the calling party to the called party, To establish a VoIP Internet phone connection between the calling party and the called party, and thereby improve the VoIP Internet phone call pass rate between the calling party and the called party.

为使本申请实施例的目的、技术方案和优点更加清楚,下面将结合本申请实施例中的附图,对本申请实施例中的技术方案进行清楚、完整地描述,显然,所描述的实施例是本申请一部分实施例,而不是全部实施例。基于本申请中的实施例,本领域普通技术人员在没有做出创造性劳动前提下所获得的所有其他实施例,都属于本申请保护的范围。In order to make the purposes, technical solutions and advantages of the embodiments of the present application clearer, the technical solutions in the embodiments of the present application will be described clearly and completely below with reference to the drawings in the embodiments of the present application. Obviously, the described embodiments It is a part of the embodiments of the present application, but not all of the embodiments. Based on the embodiments in the present application, all other embodiments obtained by those of ordinary skill in the art without creative efforts shall fall within the protection scope of the present application.

实施例一Example 1

请参见图1,图1为本公开实施例提供的一种VoIP网络电话的连接方法的流程图,具体包括以下步骤:Please refer to FIG. 1. FIG. 1 is a flowchart of a method for connecting a VoIP phone according to an embodiment of the present disclosure, which specifically includes the following steps:

步骤S11:当接收到主叫方向被叫方发送的第一呼叫请求时,向被叫方的注册地址发送第一呼叫请求,并判断第一呼叫请求是否到达被叫方;Step S11: when receiving the first call request sent by the calling party to the called party, send the first call request to the registered address of the called party, and determine whether the first call request reaches the called party;

需要说明的是,SIP服务器是一种用来提供跨越因特网的高级电话业务,利用SIP服务器可以将网络中通讯的双方建立VoIP网络电话连接。在本实施例中,是以SIP服务器为执行主体进行阐述。It should be noted that a SIP server is a kind of advanced telephony service used to provide cross-Internet, and a VoIP network telephone connection can be established between two parties communicating in the network by using the SIP server. In this embodiment, the SIP server is used as the execution subject for description.

在SIP服务器将主叫方和被叫方建立VoIP网络电话连接的过程中,如果被叫方的安卓后台没有被系统和用户杀掉,以及被叫方的NAT端口没有出现老化的情况下,那么,被叫方的当前地址会实时发送至SIP服务器。在此种情况下,SIP服务器就能够根据被叫方在SIP服务器上的注册地址找到被叫方。When the SIP server establishes a VoIP connection between the calling party and the called party, if the Android background of the called party is not killed by the system and the user, and the NAT port of the called party is not aged, then , the current address of the called party will be sent to the SIP server in real time. In this case, the SIP server can find the called party according to the registered address of the called party on the SIP server.

步骤S12:若第一呼叫请求未到达被叫方,则通知具有安卓后台的第三方设备向被叫方发送目标指令,以获取被叫方的当前地址;Step S12: if the first call request does not reach the called party, then notify the third-party device with the Android background to send the target instruction to the called party to obtain the current address of the called party;

可以理解的是,如果第一呼叫请求能够到达被叫方,说明被叫方的当前地址与注册地址一致,被叫方的安卓后台没有被系统和用户杀掉,并且,被叫方的NAT端口也没有出现老化的现象;如果第一呼叫请求不能够到达被叫方,则说明被叫方的当前地址与注册地址不一致,也即,被叫方的安卓后台被杀掉或者是被叫方的NAT端口出现了老化的现象。It is understandable that if the first call request can reach the called party, it means that the current address of the called party is the same as the registered address, the Android background of the called party has not been killed by the system and the user, and the NAT port of the called party is There is no aging phenomenon; if the first call request cannot reach the called party, it means that the current address of the called party is inconsistent with the registered address, that is, the Android background of the called party is killed or the called party's The NAT port is aging.

如果第一呼叫请求没有到达被叫方,在步骤S11的基础上,SIP服务器通知具有安卓后台的第三方设备将第一呼叫请求发送至被叫方,以使得SIP服务器可以获取到被叫方的当前地址。如图2所示,是本实施例中SIP服务器通知具有安卓后台的第三方设备呼叫被叫方的示意图。If the first call request does not reach the called party, on the basis of step S11, the SIP server notifies the third-party device with an Android background to send the first call request to the called party, so that the SIP server can obtain the called party's current address. As shown in FIG. 2 , it is a schematic diagram of the SIP server notifying a third-party device with an Android background to call the called party in this embodiment.

具体的,SIP服务器通过第三方设备中所特有的安卓通信信道向第三方设备发送目标指令,以通知第三方设备呼叫被叫方,当被叫方接收到第三方设备发送的目标指令时,被叫方就可以向SIP服务器发起地址注册请求,并将被叫方的当前地址注册在SIP服务器上,由此SIP服务器就可以获取到被叫方的当前地址。Specifically, the SIP server sends a target command to the third-party device through the Android communication channel unique to the third-party device to notify the third-party device to call the called party. When the called party receives the target command sent by the third-party device, the called party is called The calling party can initiate an address registration request to the SIP server, and register the current address of the called party on the SIP server, so that the SIP server can obtain the current address of the called party.

作为一种优选的实施方式,可以将第三方设备设置为手机,因为手机不仅可以减少将主叫方和被叫方建立VoIP网络电话的通信成本,而且,还可以提高将主叫方和被叫方建立VoIP网络电话的普适性与通用性。所以,将第三方设备设置为手机时,可以使得本申请中的方法应用于更多的实际应用场景当中。As a preferred implementation, the third-party device can be set as a mobile phone, because the mobile phone can not only reduce the communication cost of establishing a VoIP network phone between the calling party and the called party, but also improve the connection between the calling party and the called party. To establish the universality and versatility of VoIP Internet telephony. Therefore, when the third-party device is set as a mobile phone, the method in this application can be applied to more practical application scenarios.

具体的,可以将具有安卓后台的手机设置为:小米手机、华为手机和三星手机等等,在本实施例中对于手机的类型和型号不作具体限定,只要能够达到实际应用目的即可。当然,在实际应用当中,具有安卓后台的第三方设备多种多样,比如:具有安卓后台的电脑,或者是其它具有安卓后台的通信设备,所以,在本实施例中,对于第三方设备的具体类型不作具体限定。Specifically, the mobile phones with Android background can be set as: Xiaomi mobile phones, Huawei mobile phones, Samsung mobile phones, etc. In this embodiment, the type and model of the mobile phone are not specifically limited, as long as the practical application purpose can be achieved. Of course, in practical applications, there are various third-party devices with Android backgrounds, such as computers with Android backgrounds, or other communication devices with Android backgrounds. Therefore, in this embodiment, the specific The type is not specifically limited.

步骤S13:当获取到被叫方的当前地址时,则向被叫方的当前地址发送第二呼叫请求,以在主叫方和被叫方之间建立VoIP网络电话连接。Step S13: When the current address of the called party is acquired, a second call request is sent to the current address of the called party to establish a VoIP network telephone connection between the calling party and the called party.

在步骤S12的基础上,步骤S13旨在根据被叫方的当前地址将主叫方与被叫方建立VoIP网络电话连接。如图3所示,是SIP服务器重新呼叫被叫方的示意图。也即,当SIP服务器获取到了被叫方的当前地址时,SIP服务器再次向被叫方发送第二呼叫请求;此时,被叫方就能够接收到SIP服务器所发送的第二呼叫请求,并返回呼叫成功的消息给SIP服务器,SIP服务器再将呼叫成功的消息反馈至主叫方,那么,主叫方和被叫方就可以建立VoIP网络电话连接。On the basis of step S12, step S13 aims to establish a VoIP network telephone connection between the calling party and the called party according to the current address of the called party. As shown in Figure 3, it is a schematic diagram of the SIP server calling the called party again. That is, when the SIP server obtains the current address of the called party, the SIP server sends the second call request to the called party again; at this time, the called party can receive the second call request sent by the SIP server, and The call success message is returned to the SIP server, and the SIP server feeds back the call success message to the calling party. Then, the calling party and the called party can establish a VoIP network phone connection.

显然,通过本实施例中的方法,可以在被叫方由于NAT老化或者是安卓后台被杀死的情况下,也可以使得主叫方的呼叫请求发送至被叫方,由此显著提高了VoIP网络电话的呼通率。Obviously, with the method in this embodiment, the call request of the calling party can also be sent to the called party when the called party is killed due to NAT aging or the Android background, thereby significantly improving the VoIP VoIP call through rate.

此外,为了使得主叫方和被叫方快速建立VoIP网络电话连接,被叫方在接收到SIP服务器的目标指令时,被叫方还可以首先判断被叫方的VoIP应用是否处于关闭状态,如果被叫方的VoIP应用处于运行状态,则说明主叫方和被叫方的网络电话通路是处于连通状态的;如果被叫方的VoIP应用处于关闭状态,此时,被叫方可以先运行VoIP应用,这样就可以减少在将主叫方和被叫方之间建立VoIP网络电话连接的时间延迟,由此可以进一步提高主叫方和被叫方建立VoIP网络电话连接的通信效率。In addition, in order to enable the calling party and the called party to quickly establish a VoIP network phone connection, when the called party receives the target command from the SIP server, the called party can also first determine whether the called party's VoIP application is in the closed state. If the VoIP application of the called party is running, it means that the VoIP channel between the calling party and the called party is connected; if the VoIP application of the called party is closed, the called party can run the VoIP application first. In this way, the time delay for establishing the VoIP Internet phone connection between the calling party and the called party can be reduced, thereby further improving the communication efficiency of establishing the VoIP Internet phone connection between the calling party and the called party.

通过上述技术方案可知,在本实施例中当SIP服务器接收到主叫方向被叫方发送的第一呼叫请求时,首先是将第一呼叫请求发送至被叫方在SIP服务器上的注册地址,并判断第一呼叫请求是否到达被叫方,如果第一呼叫请求没有到达被叫方,则说明被叫方的NAT老化或者是安卓后台被杀死,此时,SIP服务器通知具有安卓后台的第三方设备发送目标指令,以获取被叫方的当前地址。当SIP服务器获取到被叫方的当前地址时,就可以向被叫方的当前地址发送第二呼叫请求,并以此来将主叫方和被叫方建立VoIP网络电话连接。显然,相比于现有技术中,如果被叫方出现NAT老化或者是安卓后台被杀死的情况下,SIP服务器无法将呼叫请求发送至被叫方而言,在本公开中SIP服务器是通过调用具有安卓后台的第三方设备来将主叫方的呼叫请求发送至被叫方,由此大大了提高主叫方和被叫方VoIP网络电话的呼通率。It can be seen from the above technical solutions that in this embodiment, when the SIP server receives the first call request sent by the calling party to the called party, it first sends the first call request to the registered address of the called party on the SIP server, And judge whether the first call request reaches the called party. If the first call request does not reach the called party, it means that the NAT of the called party is aging or the Android background is killed. At this time, the SIP server notifies the third party with the Android background. The third-party device sends the target command to obtain the current address of the called party. When the SIP server obtains the current address of the called party, it can send a second call request to the current address of the called party, and thereby establish a VoIP network telephone connection between the calling party and the called party. Obviously, compared with the prior art, if the called party has NAT aging or the Android background is killed, the SIP server cannot send the call request to the called party. In the present disclosure, the SIP server uses the A third-party device with an Android background is called to send the calling party's call request to the called party, thereby greatly improving the call-through rate of the calling party and the called party's VoIP network phone.

实施例二Embodiment 2

请参见图4,图4为本实施例提供的另一种VoIP网络电话的连接方法的流程图。具体包括以下步骤:Referring to FIG. 4 , FIG. 4 is a flowchart of another method for connecting a VoIP phone according to this embodiment. Specifically include the following steps:

步骤S21:当接收到主叫方向被叫方发送的第一呼叫请求时,向被叫方的注册地址发送第一呼叫请求,并判断第一呼叫请求是否到达被叫方;Step S21: when receiving the first call request sent by the calling party to the called party, send the first call request to the registered address of the called party, and determine whether the first call request reaches the called party;

步骤S22:若第一呼叫请求未到达被叫方,则通知第三方设备向被叫方发送目标指令;Step S22: if the first call request does not reach the called party, notify the third-party device to send the target instruction to the called party;

步骤S23:接收被叫方基于目标指令发送的注册请求;Step S23: receiving the registration request sent by the called party based on the target instruction;

步骤S24:在注册成功后,将被叫方新注册的地址确定为被叫方的当前地址;Step S24: after the registration is successful, the newly registered address of the called party is determined as the current address of the called party;

步骤S25:当获取到被叫方的当前地址时,则向被叫方的当前地址发送第二呼叫请求,以在主叫方和被叫方之间建立VoIP网络电话连接。Step S25: When the current address of the called party is acquired, a second call request is sent to the current address of the called party to establish a VoIP network telephone connection between the calling party and the called party.

需要说明的是,步骤S21和步骤S25与步骤11和步骤S13相同,并无改变,相关描述可参见步骤S11和步骤S13,在此不再赘述。It should be noted that, step S21 and step S25 are the same as step 11 and step S13, and there is no change, and the related description can refer to step S11 and step S13, which will not be repeated here.

在本实施例旨在说明,如果第一呼叫请求没有到达被叫方,则说明被叫方在SIP服务器上的注册地址发生了改变,也即,被叫方的注册地址当前为不可用状态,此时,SIP服务器为了获取被叫方的当前地址,是通知第三方设备向被叫方发送目标指令,以此来获取被叫方的当前地址。In this embodiment, it is intended to illustrate that if the first call request does not reach the called party, it means that the registered address of the called party on the SIP server has changed, that is, the registered address of the called party is currently unavailable, At this time, in order to obtain the current address of the called party, the SIP server notifies the third-party device to send a target instruction to the called party, so as to obtain the current address of the called party.

具体的,当被叫方接收到第三方设备发送的目标指令时,则向SIP服务器发送注册请求,当被叫方在SIP服务器上注册成功之后,SIP服务器则将被叫方之前的注册地址更新为被叫方新注册的地址,也即,SIP服务器是将被叫方新注册的地址确定为被叫方的当前地址,这样一来,SIP服务器就获取到了被叫方的当前地址。能够想到的是,当SIP服务器获取到了被叫方的当前地址之后,SIP服务器再向被叫方发送第二呼叫请求时,就可以将主叫方和被叫方建立VoIP网络电话连接。Specifically, when the called party receives the target instruction sent by the third-party device, it sends a registration request to the SIP server. After the called party registers successfully on the SIP server, the SIP server updates the previous registration address of the called party. The address newly registered for the called party, that is, the SIP server determines the newly registered address of the called party as the current address of the called party. In this way, the SIP server obtains the current address of the called party. It is conceivable that after the SIP server obtains the current address of the called party, the SIP server sends a second call request to the called party, and can establish a VoIP network telephone connection between the calling party and the called party.

可见,通过本实施例中的方法,可以保证SIP服务器获取被叫方的当前地址的可用性及可靠性。It can be seen that through the method in this embodiment, the availability and reliability of the SIP server to obtain the current address of the called party can be guaranteed.

实施例三Embodiment 3

请参见图5,图5为本实施例提供的另一种VoIP网络电话的连接方法的流程图。具体包括以下步骤:Referring to FIG. 5 , FIG. 5 is a flowchart of another method for connecting a VoIP phone according to this embodiment. Specifically include the following steps:

步骤S31:当接收到主叫方向被叫方发送的第一呼叫请求时,向被叫方的注册地址发送第一呼叫请求,并判断第一呼叫请求是否到达被叫方;Step S31: when receiving the first call request sent by the calling party to the called party, send the first call request to the registered address of the called party, and determine whether the first call request reaches the called party;

步骤S32:若第一呼叫请求未到达被叫方,则通知具有安卓后台的第三方设备的推送服务向被叫方发送目标指令,以获取被叫方的当前地址;Step S32: if the first call request does not reach the called party, then notify the push service of the third-party device with the Android background to send the target instruction to the called party to obtain the current address of the called party;

步骤S33:当获取到被叫方的当前地址时,则向被叫方的当前地址发送第二呼叫请求,以在主叫方和被叫方之间建立VoIP网络电话连接。Step S33: When the current address of the called party is acquired, a second call request is sent to the current address of the called party to establish a VoIP network telephone connection between the calling party and the called party.

需要说明的是,步骤S31和步骤S33与步骤11和步骤S13相同,并无改变,相关描述可参见步骤S11和步骤S13,在此不再赘述。It should be noted that, step S31 and step S33 are the same as step 11 and step S13, and there is no change. For the relevant description, refer to step S11 and step S13, which will not be repeated here.

本实施例旨在说明,如果第一呼叫请求没有到达被叫方时,说明被叫方的当前地址与被叫方在SIP服务器上的注册地址并不相同。在此情况下,SIP服务器可以通过调用第三方设备中的推送服务来将第一呼叫请求发送至被叫方,并以此来获取到被叫方的当前地址。The purpose of this embodiment is to illustrate that if the first call request does not reach the called party, it indicates that the current address of the called party is not the same as the registered address of the called party on the SIP server. In this case, the SIP server may send the first call request to the called party by invoking the push service in the third-party device, and thereby obtain the current address of the called party.

由于推送服务的本质是将第一呼叫请求由被动发送更改为主动发送,这样一来,就可以减少第一呼叫请求在由SIP服务器向被叫方传输信息过程中的时间,从而使得SIP服务器和被叫方的通信过程可以更为快速。Since the essence of the push service is to change the first call request from passive sending to active sending, it can reduce the time for the first call request in the process of transmitting information from the SIP server to the called party, so that the SIP server and the called party can communicate with each other. The communication process of the called party can be faster.

另外,由于第三方设备可以通过安卓后台中的C2DM(Android Cloud to DeviceMessaging)服务、XMPP(Extensible Messaging and Presence Protocol)协议、MQTT(Message Queuing Telemetry Transport)协议等通信信道来将第一呼叫请求发送至被叫方,所以,在实际应用当中,可以通过推送服务中不同的通信信道将第一呼叫请求发送至被叫方,本实施例对于通信信道的具体实现方式不作限定,只要能够达到实际应用目的即可。In addition, since the third-party device can send the first call request to the The called party, therefore, in practical applications, the first call request can be sent to the called party through different communication channels in the push service. This embodiment does not limit the specific implementation of the communication channel, as long as the actual application purpose can be achieved That's it.

可见,在本实施例中,SIP服务器通过具有安卓后台的第三方设备的推送服务来向被叫方发送目标指令,并以此来获取被叫方的当前地址,可以使得SIP服务器与被叫方的通信过程更为快速、有效。It can be seen that in this embodiment, the SIP server sends the target instruction to the called party through the push service of the third-party device with the Android background, and obtains the current address of the called party by this, so that the SIP server can communicate with the called party. The communication process is faster and more efficient.

实施例四Embodiment 4

请参见图6,图6为本实施例提供的另一种VoIP网络电话的连接方法的流程图。具体包括以下步骤:Referring to FIG. 6 , FIG. 6 is a flowchart of another method for connecting a VoIP phone according to this embodiment. Specifically include the following steps:

步骤S41:当接收到主叫方向被叫方发送的第一呼叫请求时,向被叫方的注册地址发送第一呼叫请求,并判断第一呼叫请求是否到达被叫方;Step S41: when receiving the first calling request sent by the calling party to the called party, send the first calling request to the registered address of the called party, and determine whether the first calling request reaches the called party;

步骤S42:若第一呼叫请求未到达被叫方,则通知具有安卓后台的第三方设备的短信服务向被叫方发送目标指令,以获取被叫方的当前地址;Step S42: if the first call request does not reach the called party, then notify the short message service of the third-party device with the Android background to send the target instruction to the called party to obtain the current address of the called party;

步骤S43:当获取到被叫方的当前地址时,则向被叫方的当前地址发送第二呼叫请求,以在主叫方和被叫方之间建立VoIP网络电话连接。Step S43: When the current address of the called party is acquired, a second call request is sent to the current address of the called party to establish a VoIP network telephone connection between the calling party and the called party.

需要说明的是,步骤S41和步骤S43与步骤11和步骤S13相同,并无改变,相关描述可参见步骤S41和步骤S43,在此不再赘述。It should be noted that, step S41 and step S43 are the same as step 11 and step S13 without any change, and the related description can refer to step S41 and step S43, which will not be repeated here.

本实施例旨在说明,在实际应用当中,SIP服务器除了可以通过第三方设备的推送服务向被叫方发送目标指令之外,还可以通过第三方设备的短信服务向被叫方发送目标指令,来获取被叫方的当前地址。This embodiment is intended to illustrate that, in practical applications, the SIP server can not only send target instructions to the called party through the push service of the third-party device, but also send the target instructions to the called party through the short message service of the third-party device, to get the current address of the called party.

因为短信服务不仅成本低廉,而且,短信服务还能够清楚、直观的显示通信内容,所以,通过第三方设备的短信服务来向被叫方发送目标指令,不仅可以减少第三方设备的通信成本,而且,还可以进一步提高用户体验。此外,短信服务还具有网络覆盖范围广、移动性能强的优点,所以,通过第三方设备的短信服务向来获取被叫方的当前地址时,还可以提高SIP服务器在获取被叫方的当前地址时的整体性能和获取速度。Because the short message service is not only low-cost, but also can clearly and intuitively display the communication content, so sending the target command to the called party through the short message service of the third-party device can not only reduce the communication cost of the third-party device, but also , which can further improve the user experience. In addition, the short message service also has the advantages of wide network coverage and strong mobile performance. Therefore, when the current address of the called party is obtained through the short message service of the third-party device, it can also improve the SIP server when obtaining the current address of the called party. overall performance and acquisition speed.

可见,在本实施例中,SIP服务器通过具有安卓后台的第三方设备的短信服务来向被叫方发送目标指令,并以此来获取被叫方的当前地址,不仅可以降低第三方设备的通信成本,而且,也可以保证SIP服务器在获取被叫方的当前地址时的整体性能。It can be seen that, in this embodiment, the SIP server sends the target instruction to the called party through the short message service of the third-party device with the Android background, and uses this to obtain the current address of the called party, which can not only reduce the communication of the third-party device cost, and also can guarantee the overall performance of the SIP server in obtaining the current address of the called party.

实施例五Embodiment 5

请参见图7,图7为本实施例提供的另一种VoIP网络电话的连接方法的流程图。具体包括以下步骤:Referring to FIG. 7 , FIG. 7 is a flowchart of another method for connecting a VoIP phone according to this embodiment. Specifically include the following steps:

步骤S51:当接收到主叫方向被叫方发送的第一呼叫请求时,向被叫方的注册地址发送第一呼叫请求,并判断第一呼叫请求是否到达被叫方;Step S51: when receiving the first call request sent by the calling party to the called party, send the first call request to the registered address of the called party, and determine whether the first call request reaches the called party;

步骤S52:若第一呼叫请求未到达被叫方,则通知具有安卓后台的第三方设备向被叫方发送目标指令;Step S52: if the first call request does not reach the called party, then notify the third-party device with the Android background to send the target instruction to the called party;

步骤S53:判断是否在预设时长内获取到被叫方的当前地址;Step S53: judging whether the current address of the called party is obtained within a preset duration;

步骤S54:若未在预设时长内获取到当前地址,则向主叫方返回呼叫失败信息。Step S54: If the current address is not obtained within the preset time period, return call failure information to the calling party.

步骤S51和步骤S52与步骤S11和步骤S12相同,并无改变,相关描述信息可参见步骤S11和步骤S12,在此不再赘述。Step S51 and step S52 are the same as step S11 and step S12 without any change. For the relevant description information, please refer to step S11 and step S12, which will not be repeated here.

本实施例旨在说明,如果SIP服务器不能够在预设时长内获取到被叫方的当前地址,则说明被叫方出现了异常,此时,SIP服务器向主叫方返回呼叫失败信息。当主叫方接收到SIP服务器返回的呼叫失败信息以后,则可以根据实际情况的需要,决定主叫方是继续向SIP服务器发送目标指令,来获取被叫方的当前地址,还是结束操作流程,这样就可以避免主叫方长时间对SIP服务器的监听,而对主叫方通信资源所造成的浪费。The purpose of this embodiment is to illustrate that if the SIP server cannot obtain the current address of the called party within a preset time period, it means that the called party is abnormal, and at this time, the SIP server returns call failure information to the calling party. When the calling party receives the call failure information returned by the SIP server, it can decide whether the calling party should continue to send the target command to the SIP server to obtain the current address of the called party or end the operation process according to the actual situation. In this way, the calling party can avoid the waste of the calling party's communication resources caused by monitoring the SIP server for a long time.

作为一种优选的实施方式,可以将预设时长设置为500ms,这样可以避免主叫方在获取被叫方的当前地址过程中,主叫方等待时间过长,而影响主叫方的用户体验。As a preferred implementation, the preset duration can be set to 500ms, which can prevent the calling party from waiting too long during the process of obtaining the current address of the called party, which will affect the user experience of the calling party .

可见,SIP服务器如果不能在预设时长内获取到被叫方的当前地址,则向主叫方返回呼叫失败信息,可以避免主叫方长时间监听SIP服务器,而对主叫方通信资源的浪费,并由此提高了主叫方的整体通信效率。It can be seen that if the SIP server cannot obtain the current address of the called party within the preset time period, it will return the call failure information to the calling party, which can prevent the calling party from monitoring the SIP server for a long time and waste the communication resources of the calling party. , and thereby improve the overall communication efficiency of the calling party.

实施例六Embodiment 6

请参见图8,图8为本实施例提供的一种VoIP网络电话的连接方法,该方法应用于被叫方。具体包括以下步骤:Please refer to FIG. 8 . FIG. 8 provides a method for connecting a VoIP internet phone according to this embodiment, and the method is applied to the called party. Specifically include the following steps:

步骤S61:接收独立于SIP服务器的具有安卓后台的第三方设备发送的目标指令,目标指令用于指示SIP服务器向被叫方发送主叫方的呼叫请求;Step S61: receiving a target instruction sent by a third-party device with an Android background that is independent of the SIP server, and the target instruction is used to instruct the SIP server to send the calling party's call request to the called party;

步骤S62:根据目标指令,运行VoIP应用;Step S62: according to the target instruction, run the VoIP application;

步骤S63:向SIP服务器发送注册请求,注册新的当前地址,以使SIP服务器向当前地址发送主叫方的呼叫请求。Step S63: Send a registration request to the SIP server to register a new current address, so that the SIP server sends a call request of the calling party to the current address.

在本实施例中,是以被叫方为执行主体对VoIP网络电话的连接方法进行说明。可以理解的是,如果被叫方的安卓后台被系统和用户杀掉,或者是被叫方的NAT端口出现老化,则主叫方的呼叫请求无法到达被叫方。In this embodiment, the method for connecting a VoIP phone is described with the called party as the execution subject. It is understandable that if the Android background of the called party is killed by the system and the user, or the NAT port of the called party is aging, the call request of the calling party cannot reach the called party.

但是,如果SIP服务器是通过具有安卓后台的第三方设备来向被叫方发送目标指令,那么,主叫方的呼叫请求就能够到达被叫方。此时,被叫方就能够根据目标指令,运行VoIP应用,并向SIP服务器发送注册请求,然后,被叫方的当前地址就能够注册到SIP服务器当中。However, if the SIP server sends the target instruction to the called party through a third-party device with an Android background, then the call request of the calling party can reach the called party. At this point, the called party can run the VoIP application according to the target instruction and send a registration request to the SIP server, and then the current address of the called party can be registered with the SIP server.

能够想到的是,当被叫方的当前地址注册到SIP服务器当中时,SIP服务器就能够根据被叫方注册的当前地址,将主叫方的呼叫请求发送至被叫方,当主叫方的呼叫请求到达被叫方时,主叫方和被叫方就可以建立VoIP网络电话连接。It is conceivable that when the current address of the called party is registered with the SIP server, the SIP server can send the call request of the calling party to the called party according to the current address registered by the called party. When the call request arrives at the called party, the calling party and the called party can establish a VoIP network telephone connection.

显然,相比于现有技术中,如果被叫方出现NAT老化或者是安卓后台被杀死的情况下,SIP服务器无法将呼叫请求发送至被叫方而言,在本公开中SIP服务器是通过调用具有安卓后台的第三方设备来将主叫方的呼叫请求发送至被叫方,由此便大大了提高主叫方和被叫方VoIP网络电话的呼通率。Obviously, compared with the prior art, if the called party has NAT aging or the Android background is killed, the SIP server cannot send the call request to the called party. In the present disclosure, the SIP server uses the A third-party device with an Android background is called to send the call request of the calling party to the called party, thereby greatly improving the call-through rate of the VoIP network phone between the calling party and the called party.

实施例七Embodiment 7

请参见图9,图9为本公开实施例提供的一种VoIP网络电话的连接装置的结构框图,该连接装置应用于SIP服务器,该连接装置包括:Please refer to FIG. 9. FIG. 9 is a structural block diagram of a connection device for a VoIP phone according to an embodiment of the present disclosure. The connection device is applied to a SIP server, and the connection device includes:

请求发送模块101,用于当接收到主叫方向被叫方发送的第一呼叫请求时,向被叫方的注册地址发送第一呼叫请求,并判断第一呼叫请求是否到达被叫方;The request sending module 101 is configured to send the first call request to the registered address of the called party when receiving the first call request sent by the calling party to the called party, and determine whether the first call request reaches the called party;

地址获取模块102,用于若第一呼叫请求未到达被叫方,则通知具有安卓后台的第三方设备向被叫方发送目标指令,以获取被叫方的当前地址;The address obtaining module 102 is configured to notify the third-party device with the Android background to send the target instruction to the called party to obtain the current address of the called party if the first call request does not reach the called party;

电话建立模块103,用于当获取到被叫方的当前地址时,则向被叫方的当前地址发送第二呼叫请求,以在主叫方和被叫方之间建立VoIP网络电话连接。The phone establishing module 103 is configured to send a second call request to the current address of the called party when the current address of the called party is obtained, so as to establish a VoIP network telephone connection between the calling party and the called party.

其中,上述地址获取模块102可以包括:Wherein, the above-mentioned address obtaining module 102 may include:

指令发送单元,用于通知第三方设备向被叫方发送目标指令;an instruction sending unit, used to notify the third-party device to send the target instruction to the called party;

请求接收单元,用于接收被叫方基于目标指令发送的注册请求;a request receiving unit, configured to receive a registration request sent by the called party based on the target instruction;

地址更新单元,用于在注册成功后,将被叫方新注册的地址确定为被叫方的当前地址。The address updating unit is used to determine the newly registered address of the called party as the current address of the called party after the registration is successful.

其中,上述地址获取模块102可以包括:Wherein, the above-mentioned address obtaining module 102 may include:

第一指令发送单元,用于通知具有安卓后台的第三方设备的推送服务向被叫方发送目标指令。The first instruction sending unit is used to notify the push service of the third-party device with the Android background to send the target instruction to the called party.

其中,上述地址获取模块102可以包括:Wherein, the above-mentioned address obtaining module 102 may include:

第二指令发送单元,用于通知具有安卓后台的第三方设备的短信服务向被叫方发送目标指令。The second instruction sending unit is used to notify the short message service of the third-party device with the Android background to send the target instruction to the called party.

其中,上述VoIP网络电话的连接装置还包括:Wherein, the connection device of the above-mentioned VoIP Internet phone also includes:

时长判断模块,用于在通知具有安卓后台的第三方设备向被叫方发送目标指令之后,判断是否在预设时长内获取到被叫方的当前地址;The duration judgment module is used to judge whether the current address of the called party is obtained within the preset duration after notifying the third-party device with the Android background to send the target command to the called party;

信息返回模块,用于若未在预设时长内获取到当前地址,则向主叫方返回呼叫失败信息。The information return module is used to return the call failure information to the calling party if the current address is not obtained within the preset time period.

请参见图10,图10为本公开实施例提供的另一种VoIP网络电话的连接装置的结构框图,该连接装置应用于被叫方,该连接装置包括:Please refer to FIG. 10. FIG. 10 is a structural block diagram of another VoIP internet phone connection device provided by an embodiment of the present disclosure. The connection device is applied to the called party, and the connection device includes:

指令接收模块201,用于接收独立于SIP服务器的第三方设备发送的目标指令,目标指令用于指示SIP服务器向被叫方发送主叫方的呼叫请求;The instruction receiving module 201 is used to receive a target instruction sent by a third-party device independent of the SIP server, and the target instruction is used to instruct the SIP server to send the calling party's call request to the called party;

应用运行模块202,用于根据目标指令,运行VoIP应用;an application running module 202, configured to run the VoIP application according to the target instruction;

地址注册模块203,用于向SIP服务器发送注册请求,注册新的当前地址,以使SIP服务器向当前地址发送主叫方的呼叫请求。The address registration module 203 is configured to send a registration request to the SIP server to register a new current address, so that the SIP server sends a call request of the calling party to the current address.

上述实施例中的装置,其中各个模块执行操作的具体方式已经在有关该方法的实施例中进行了详细描述,此处将不做详细说明。In the apparatus in the above embodiment, the specific manner in which each module performs operations has been described in detail in the embodiments related to the method, and will not be described in detail here.

图11是根据一示例性实施例示出的一种VoIP网络电话的连接设备300的框图。如图8所示,该VoIP网络电话的连接设备300可以包括:处理器301,存储器302。该VoIP网络电话的连接设备300还可以包括多媒体组件303,信息输入/信息输出(I/O)接口304,以及通信组件305中的一者或多者。FIG. 11 is a block diagram of a connection device 300 of a VoIP internet phone according to an exemplary embodiment. As shown in FIG. 8 , the connection device 300 of the VoIP internet phone may include: a processor 301 and a memory 302 . The VoIP internet phone connection device 300 may also include one or more of a multimedia component 303 , an information input/information output (I/O) interface 304 , and a communication component 305 .

其中,处理器301用于控制该电子设备300的整体操作,以完成上述的应用于VoIP网络电话的连接方法中的全部或部分步骤;存储器302用于存储各种类型的数据以支持在该VoIP网络电话的连接设备300的操作,这些数据例如可以包括用于在该VoIP网络电话的连接设备300上操作的任何应用程序或方法的指令,以及应用程序相关的数据,例如联系人数据、收发的消息、图片、音频、视频等等。该存储器302可以由任何类型的易失性或非易失性存储设备或者它们的组合实现,例如静态随机存取存储器(Static Random AccessMemory,简称SRAM),电可擦除可编程只读存储器(Electrically Erasable ProgrammableRead-Only Memory,简称EEPROM),可擦除可编程只读存储器(Erasable ProgrammableRead-Only Memory,简称EPROM),可编程只读存储器(Programmable Read-Only Memory,简称PROM),只读存储器(Read-Only Memory,简称ROM),磁存储器,快闪存储器,磁盘或光盘。Wherein, the processor 301 is used to control the overall operation of the electronic device 300 to complete all or part of the steps in the above-mentioned connection method applied to the VoIP Internet phone; the memory 302 is used to store various types of data to support the VoIP The operation of the connection device 300 of the VoIP phone, for example, these data may include instructions for any application or method operating on the connection device 300 of the VoIP Internet phone, as well as application-related data, such as contact data, transceived Messages, pictures, audio, video, and more. The memory 302 can be implemented by any type of volatile or non-volatile memory device or a combination thereof, such as static random access memory (Static Random Access Memory, SRAM for short), electrically erasable programmable read-only memory (Electrically Erasable Programmable Read Only Memory) Erasable Programmable Read-Only Memory, EEPROM for short), Erasable Programmable Read-Only Memory (EPROM), Programmable Read-Only Memory (PROM), Read Only Memory (Read -Only Memory, referred to as ROM), magnetic memory, flash memory, magnetic disk or optical disk.

多媒体组件303可以包括屏幕和音频组件。其中屏幕例如可以是触摸屏,音频组件用于输出和/或输入音频信号。例如,音频组件可以包括一个麦克风,麦克风用于接收外部音频信号。所接收的音频信号可以被进一步存储在存储器302或通过通信组件305发送。音频组件还包括至少一个扬声器,用于输出音频信号。I/O接口304为处理器301和其他接口模块之间提供接口,上述其他接口模块可以是键盘,鼠标,按钮等。这些按钮可以是虚拟按钮或者实体按钮。通信组件305用于该电子设备300与其他设备之间进行有线或无线通信。无线通信,例如Wi-Fi,蓝牙,近场通信(Near Field Communication,简称NFC),2G、3G或4G,或它们中的一种或几种的组合,因此相应的该通信组件305可以包括:Wi-Fi模块,蓝牙模块,NFC模块。Multimedia components 303 may include screen and audio components. Wherein the screen can be, for example, a touch screen, and the audio component is used for outputting and/or inputting audio signals. For example, the audio component may include a microphone for receiving external audio signals. The received audio signal may be further stored in memory 302 or transmitted through communication component 305 . The audio assembly also includes at least one speaker for outputting audio signals. The I/O interface 304 provides an interface between the processor 301 and other interface modules, and the above-mentioned other interface modules may be a keyboard, a mouse, a button, and the like. These buttons can be virtual buttons or physical buttons. The communication component 305 is used for wired or wireless communication between the electronic device 300 and other devices. Wireless communication, such as Wi-Fi, Bluetooth, Near Field Communication (NFC for short), 2G, 3G or 4G, or one or a combination of them, so the corresponding communication component 305 may include: Wi-Fi module, Bluetooth module, NFC module.

在一示例性实施例中,VoIP网络电话的连接设备300可以被一个或多个应用专用集成电路(Application Specific Integrated Circuit,简称ASIC)、数字信号处理器(Digital Signal Processor,简称DSP)、数字信号处理设备(Digital Signal ProcessingDevice,简称DSPD)、可编程逻辑器件(Programmable Logic Device,简称PLD)、现场可编程门阵列(Field Programmable Gate Array,简称FPGA)、控制器、微控制器、微处理器或其他电子元件实现,用于执行上述给出的VoIP网络电话的连接方法。In an exemplary embodiment, the connection device 300 of the VoIP internet phone may be implemented by one or more application specific integrated circuits (Application Specific Integrated Circuit, ASIC for short), digital signal processors (Digital Signal Processor, DSP for short), digital signal Digital Signal Processing Device (DSPD), Programmable Logic Device (PLD), Field Programmable Gate Array (FPGA), controller, microcontroller, microprocessor or Other electronic components are implemented to implement the connection method of the VoIP internet phone given above.

在另一示例性实施例中,还提供了一种包括程序指令的计算机可读存储介质,该程序指令被处理器执行时实现上述VoIP网络电话的连接方法的步骤。例如,该计算机可读存储介质可以为上述存储有程序指令的存储器302,上述程序指令可由VoIP网络电话的连接设备300的处理器301执行以完成上述VoIP网络电话的连接方法。In another exemplary embodiment, a computer-readable storage medium including program instructions is also provided, and when the program instructions are executed by a processor, the steps of the above method for connecting a VoIP internet phone are implemented. For example, the computer-readable storage medium can be the above-mentioned memory 302 storing program instructions, and the above-mentioned program instructions can be executed by the processor 301 of the VoIP Internet phone connection device 300 to complete the above-mentioned VoIP Internet phone connection method.

Claims (10)

1.一种VoIP网络电话的连接方法,其特征在于,应用于SIP服务器,包括:1. the connection method of a VoIP internet phone, is characterized in that, is applied to SIP server, comprises: 当接收到主叫方向被叫方发送的第一呼叫请求时,向所述被叫方的注册地址发送所述第一呼叫请求,并判断所述第一呼叫请求是否到达所述被叫方;When receiving the first call request sent by the calling party to the called party, send the first call request to the registered address of the called party, and determine whether the first call request reaches the called party; 若所述第一呼叫请求未到达所述被叫方,则通知具有安卓后台的第三方设备向所述被叫方发送目标指令,以获取所述被叫方的当前地址;If the first call request does not reach the called party, notifying a third-party device with an Android background to send a target instruction to the called party to obtain the current address of the called party; 当获取到所述被叫方的当前地址时,则向所述被叫方的当前地址发送第二呼叫请求,以在所述主叫方和所述被叫方之间建立VoIP网络电话连接。When the current address of the called party is acquired, a second call request is sent to the current address of the called party, so as to establish a VoIP network telephone connection between the calling party and the called party. 2.根据权利要求1所述的方法,其特征在于,所述通知具有安卓后台的第三方设备向所述被叫方发送目标指令,以获取所述被叫方的当前地址,包括:2 . The method according to claim 1 , wherein the notification that a third-party device with an Android background sends a target instruction to the called party to obtain the current address of the called party, comprising: 2 . 通知所述第三方设备向所述被叫方发送所述目标指令;Notifying the third-party device to send the target instruction to the called party; 接收所述被叫方基于所述目标指令发送的注册请求;receiving a registration request sent by the called party based on the target instruction; 在注册成功后,将所述被叫方新注册的地址确定为所述被叫方的当前地址。After the registration is successful, the newly registered address of the called party is determined as the current address of the called party. 3.根据权利要求1或2所述的方法,其特征在于,所述通知具有安卓后台的第三方设备向所述被叫方发送目标指令,包括:3. The method according to claim 1 or 2, wherein the notification that a third-party device with an Android background sends a target instruction to the called party, comprising: 通知具有安卓后台的所述第三方设备的推送服务向所述被叫方发送所述目标指令。Notifying the push service of the third-party device with the Android background to send the target instruction to the called party. 4.根据权利要求1或2所述的方法,其特征在于,所述通知具有安卓后台的第三方设备向所述被叫方发送目标指令,包括:4. The method according to claim 1 or 2, wherein the notification that a third-party device with an Android background sends a target instruction to the called party, comprising: 通知具有安卓后台的所述第三方设备的短信服务向所述被叫方发送所述目标指令。Notifying the short message service of the third-party device with the Android background to send the target instruction to the called party. 5.根据权利要求1或2所述的方法,其特征在于,在所述通知具有安卓后台的第三方设备向所述被叫方发送目标指令之后,还包括:5. The method according to claim 1 or 2, characterized in that, after notifying that a third-party device with an Android background sends a target instruction to the called party, the method further comprises: 判断是否在预设时长内获取到所述被叫方的当前地址;Determine whether the current address of the called party is obtained within a preset time period; 若未在所述预设时长内获取到所述当前地址,则向所述主叫方返回呼叫失败信息。If the current address is not obtained within the preset time period, call failure information is returned to the calling party. 6.一种VoIP网络电话的连接方法,其特征在于,应用于被叫方,包括:6. a connection method of VoIP internet phone, is characterized in that, is applied to the called party, comprising: 接收独立于SIP服务器的具有安卓后台的第三方设备发送的目标指令,所述目标指令用于指示所述SIP服务器向所述被叫方发送主叫方的呼叫请求;Receive a target instruction sent by a third-party device with an Android background independent of the SIP server, where the target instruction is used to instruct the SIP server to send a call request of the calling party to the called party; 根据所述目标指令,运行VoIP应用;According to the target instruction, run the VoIP application; 向所述SIP服务器发送注册请求,注册新的当前地址,以使所述SIP服务器向所述当前地址发送所述主叫方的呼叫请求。A registration request is sent to the SIP server to register a new current address, so that the SIP server sends a call request of the calling party to the current address. 7.一种VoIP网络电话的连接装置,其特征在于,应用于SIP服务器,包括:7. a connection device of VoIP network phone, is characterized in that, is applied to SIP server, comprises: 请求发送模块,用于当接收到主叫方向被叫方发送的第一呼叫请求时,向所述被叫方的注册地址发送所述第一呼叫请求,并判断所述第一呼叫请求是否到达所述被叫方;A request sending module, configured to send the first call request to the registered address of the called party when receiving the first call request sent by the calling party to the called party, and determine whether the first call request arrives the called party; 地址获取模块,用于若所述第一呼叫请求未到达所述被叫方,则通知具有安卓后台的第三方设备向所述被叫方发送目标指令,以获取所述被叫方的当前地址;The address obtaining module is configured to notify the third-party device with an Android background to send a target instruction to the called party to obtain the current address of the called party if the first call request does not reach the called party ; 电话建立模块,用于当获取到所述被叫方的当前地址时,则向所述被叫方的当前地址发送第二呼叫请求,以在所述主叫方和所述被叫方之间建立VoIP网络电话连接。The phone establishment module is configured to send a second call request to the current address of the called party when the current address of the called party is obtained, so as to establish a connection between the calling party and the called party Establish a VoIP internet phone connection. 8.一种VoIP网络电话的连接装置,其特征在于,应用于被叫方,包括:8. A connection device for a VoIP internet phone, characterized in that, applied to the called party, comprising: 指令接收模块,用于接收独立于SIP服务器的具有安卓后台的第三方设备发送的目标指令,所述目标指令用于指示所述SIP服务器向所述被叫方发送主叫方的呼叫请求;an instruction receiving module, configured to receive a target instruction sent by a third-party device with an Android background independent of the SIP server, where the target instruction is used to instruct the SIP server to send a call request of the calling party to the called party; 应用运行模块,用于根据所述目标指令,运行VoIP应用;an application running module for running the VoIP application according to the target instruction; 地址注册模块,用于向所述SIP服务器发送注册请求,注册新的当前地址,以使所述SIP服务器向所述当前地址发送所述主叫方的呼叫请求。The address registration module is configured to send a registration request to the SIP server to register a new current address, so that the SIP server sends a call request of the calling party to the current address. 9.一种VoIP网络电话的连接设备,其特征在于,包括:9. A connection device of a VoIP internet phone, characterized in that, comprising: 存储器,用于存储计算机程序;memory for storing computer programs; 处理器,用于执行所述计算机程序时实现如权利要求1至5任一项或权利要求6所述的一种VoIP网络电话的连接方法的步骤。The processor is configured to implement the steps of a VoIP internet phone connection method according to any one of claims 1 to 5 or claim 6 when executing the computer program. 10.一种计算机可读存储介质,其特征在于,所述计算机可读存储介质上存储有计算机程序,所述计算机程序被处理器执行时实现如权利要求1至5任一项或权利要求6所述的VoIP网络电话的连接方法的步骤。10. A computer-readable storage medium, characterized in that, a computer program is stored on the computer-readable storage medium, and when the computer program is executed by a processor, any one of claims 1 to 5 or claim 6 is implemented The steps of the connection method of the VoIP internet phone.
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