WO2014040433A1 - Implementation method and implementation system for ptt call based on voip technology - Google Patents

Implementation method and implementation system for ptt call based on voip technology Download PDF

Info

Publication number
WO2014040433A1
WO2014040433A1 PCT/CN2013/077159 CN2013077159W WO2014040433A1 WO 2014040433 A1 WO2014040433 A1 WO 2014040433A1 CN 2013077159 W CN2013077159 W CN 2013077159W WO 2014040433 A1 WO2014040433 A1 WO 2014040433A1
Authority
WO
WIPO (PCT)
Prior art keywords
ptt
source identifier
ptt client
voice data
client
Prior art date
Application number
PCT/CN2013/077159
Other languages
French (fr)
Chinese (zh)
Inventor
吴奇峰
Original Assignee
惠州Tcl移动通信有限公司
Priority date (The priority date is an assumption and is not a legal conclusion. Google has not performed a legal analysis and makes no representation as to the accuracy of the date listed.)
Filing date
Publication date
Application filed by 惠州Tcl移动通信有限公司 filed Critical 惠州Tcl移动通信有限公司
Publication of WO2014040433A1 publication Critical patent/WO2014040433A1/en

Links

Classifications

    • 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
    • H04L65/40Support for services or applications
    • H04L65/4061Push-to services, e.g. push-to-talk or push-to-video
    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04WWIRELESS COMMUNICATION NETWORKS
    • H04W4/00Services specially adapted for wireless communication networks; Facilities therefor
    • H04W4/06Selective distribution of broadcast services, e.g. multimedia broadcast multicast service [MBMS]; Services to user groups; One-way selective calling services
    • H04W4/10Push-to-Talk [PTT] or Push-On-Call services

Definitions

  • the invention relates to the application field of mobile terminal communication technology, in particular to a PTT call implementation method and implementation system based on VOIP technology.
  • VoIP Voice On IP
  • IP Internet-based call. From the perspective of usage, it can achieve peer-to-peer calls like a normal mobile phone.
  • the biggest advantage of VoIP is that the call is cheap, but one-to-many calls cannot be achieved.
  • PTT Push To Talk
  • the full name of PTT is Push To Talk, which is a one-touch call, which functions like a traditional walkie-talkie.
  • the advantage of PTT is that it has a group call function. It can be grouped into groups, for example, if there are ten people set up in a group, then one person is speaking and the other nine can hear it. Companies, organizations (such as hotels, airports) or organizations (such as construction teams) can form a group to use this group call function, and do not have to call one by one.
  • the technical problem to be solved by the present invention is to provide a basis for the above-mentioned drawbacks of the prior art.
  • the PTT call implementation method and implementation system of VOIP technology solves the group call function of VOIP, and makes the design of PTT group call based on VOIP technology simple and easy to promote.
  • one technical solution adopted by the present invention is to provide a method for implementing a PTT call based on VOIP technology, including the following steps: A. Heading a RTP voice data packet to be sent by a PTT client having a VOIP function Expanding, setting a data source identifier of the current call in the header extension, and transmitting the header-expanded RTP voice data packet to the PTIP-enabled PTT server; wherein the step A includes: the PTT client is to be sent The extended bit position 1 of the RTP fixed header in the RTP voice data packet increases the header extension portion for the RTP fixed header; B.
  • the PTT server reads the data of the current call in the RTP voice data packet after the header extension Source identifier, and determining, according to the range of the interval in which the value of the data source identifier of the current call is located, determining the PTT client
  • the step B further includes: when the value of the data source identifier of the current call is 0x1, the PTT client has a preemption right; when the value of the data source identifier of the current call is in the interval 0x2-0xFFFFFF, the PTT client does not have a preemption right, and applies for obtaining a floor when no third party PTT client speaks; when the value of the data source identifier of the current call is in the interval Ox 1000000-OxFFFFFFFF, The PTT client has no floor.
  • C When it is determined that the PTT client has a floor, the PTT server sends the extended RTP voice packet to the third-party PTT client in multicast form.
  • the step B further includes: when the PTT server receives the extended RTP voice data packet, determining whether there is voice data in the extended RTP voice data packet; when the header is expanded When there is no voice data in the RTP voice data packet, it is determined that the PTT client gives up the floor, otherwise the speaking permission of the PTT client is determined by the range of the range in which the value of the data source identifier of the current call is located.
  • the data source identifier of the current call in the step A uniquely identifies a PTT client, and the PTT server determines the speaking permission of the PTT client accordingly.
  • the data source identifier of the current call is 4 bytes in length.
  • another technical solution adopted by the present invention is to provide a method for implementing a PTT call based on VOIP technology, which includes the following steps: A. RTP voice data to be sent by a PTT client having a VOIP function The packet is extended by the header, the data source identifier of the current call is set in the header extension part, and the extended RTP voice data packet is sent to the POT server with the VOIP function; B. After the PTT server reads the header extension The data source identifier of the current call in the RTP voice data packet, and determining the speaking permission of the PTT client according to the range of the range in which the value of the data source identifier of the current call is located; C. determining that the PTT client has The floor, the PTT server sends the extended RTP voice data packet to the third-party PTT client in multicast form.
  • the step A includes: the PTT client adds the extended bit position 1 of the RTP fixed header in the RTP voice data packet to be sent, and adds the header extension to the RTP fixed header.
  • the step B further includes: when the value of the data source identifier of the current call is 0x1, the PTT client has a preemption right; when the value of the data source identifier of the current call is in the interval 0x2-0xFFFFFF The PTT client does not have the right to speak, and applies for obtaining the floor when no third-party PTT client speaks; when the value of the data source identifier of the current call is in the interval Ox 1000000-OxFFFFFFFF, the PTT client There is no right to speak. ⁇ .
  • the step B further includes: when the PTT server receives the extended RTP voice data packet, determining whether there is voice data in the extended RTP voice data packet; When there is no voice data in the extended RTP voice data packet, it is determined that the PTT client gives up the floor, otherwise the speaking permission of the PTT client is determined by the range of the range in which the value of the data source identifier of the current call is located.
  • the data source identifier of the current call in the step A uniquely identifies a PTT client, and the PTT server determines the speaking permission of the PTT client accordingly.
  • the data source identifier of the current call is 4 bytes in length.
  • a PTT call implementation system based on VOIP technology, which includes a PTT client and a PTT server having a VOIP function, wherein the PTT client is used for Performing header expansion on the RTP voice data packet, setting a data source identifier of the current call in the header extension portion, and transmitting the header extended RTP voice data packet to the PTT server; the PTT server is configured to receive the PTT client Sending the extended RTP voice data packet, and obtaining the data source identifier of the current call, determining the speaking permission of the PTT client by using the data source identifier of the current call, and according to the speaking permission of the PTT client Make an operation of whether to send the RTP voice data packet to the third-party PTT client in a multicast manner.
  • the PTT client and the PTT server are both VOIP-enabled mobile terminals, and the PTT client is multiple, and the PTT server is one.
  • the method and the implementation system for implementing the PTT call based on the VOIP technology provided by the present invention, by performing header expansion on the RTP voice data packet sent by the PTT client, setting the data source identifier of the current call in the header extension portion, so that the PTT server can pass the current pass.
  • the data source identifier determines the speaking permission of the PTT client, and the RTP voice data packet sent by the PTT client with the floor is sent to the third-party PTT client in multicast form to implement the group call function.
  • the implementation of the header extension of the RTP voice data packet is relatively simple, and is easy to implement.
  • the PTT server can make a correct decision only through a data source identifier, which facilitates the design and promotion of the system.
  • FIG. 1 is a schematic structural diagram of a PTT call implementation system based on VOIP technology provided by the present invention.
  • 2 is a flow chart of a method for implementing a PTT call based on VOIP technology provided by the present invention.
  • FIG. 3 is a flowchart of a method for determining a speaking right of a PTT client by a PTT server according to a VOIP-based PTT call implementation method provided by the present invention.
  • FIG. 4 is a flow chart of a method for determining a PTT server in a PTT call implementation method based on VOIP technology provided by the present invention.
  • the present invention provides a method and an implementation system for implementing a PTT call based on VOIP technology.
  • the system includes a plurality of PTT clients and a PTT server, and both the PTT client and the PTT server have VOIP (based on the Internet).
  • the mobile terminal of the function of the call, the PTT client can be a mobile phone used by the user for the user to make a PTT call, and in order to improve the data processing rate, the PTT server can be used for the computer, for voice receiving and multicasting, and which PTT is decided.
  • the client has the right to speak.
  • the system is implemented by software.
  • Each PTT client performs data transmission with the PTT server through RTP (Real-Time Transport Protocol), and the PTT server arbitrates the RTP voice data packet sent by the PTT client according to the data source identifier in the RTP voice data packet.
  • the PTT client's speaking permission is determined, and the RTP voice data packet is distributed to the third-party PTT client in a multicast form according to the speaking permission of the PTT client.
  • the PTT client is used for header expansion of the RTP voice data packet, adding an extension part after the RTP fixed header of the RTP voice data packet, and setting a data source identifier of the current call in the header extension part, and then
  • the extended RTP voice data packet is sent to the PTT server, where the data source identifier of the current call is used to identify the speaking permission of the current call of the PTT client, and the PTT server may use the value of the data source identifier of the current call. Determine the speaking permission of the PTT client.
  • the PTT server is configured to receive a header extended RTP voice data packet sent by the PTT client, obtain a data source identifier of the current call, and determine, by using a data source identifier of the current call, a speech of the PTT client. Authorization, and according to the speaking permission of the PTT client, whether to send the RTP voice data packet to the third-party PTT client in a multicast manner.
  • the present invention further provides a PTIP-based PTT call implementation method.
  • the method includes the following steps: Step S100: Passing a POT with VOIP function
  • the client performs header expansion on the RTP voice data packet to be sent, sets a data source identifier of the current call in the header extension portion, and sends the extended RTP voice data packet to the POT server with the VOIP function;
  • Step S200 The PTT server reads the data source identifier of the current call in the extended RTP voice data packet, and judges according to the range of the value of the data source identifier of the current call. Disconnecting the speaking permission of the PTT client;
  • Step S300 When it is determined that the PTT client has a floor, the PTT server sends the extended RTP voice data packet to a third-party PTT client in multicast form.
  • multiple PTT clients will simultaneously send RTP voice packets to the PTT server through the RTP protocol, and the PTT server will decide on each RTP voice packet received, and decide which PTT client has the floor, and The RTP voice data packet of the PTT client with the voice is sent to the third-party PTT client in multicast form, thereby implementing the group call function.
  • the PTT client performs header extension on the RTP voice packet to be sent.
  • the format of the RTP fixed header is as follows:
  • each data source has a data source identifier, and there is a data source set up.
  • X represents the extended bit. If X is set to 1, it means that there is an extended part after the fixed header.
  • the format of the header extension is:
  • header extension includes a 16-bit length field indicating the number of 32-bit words in the extension. Only one header extension is allowed after the RTP fixed header. The first 16 bits of the extension are used to identify the identifier or parameter. This 16-bit format is not specified in the RTP, but is defined by the upper layer protocol of the specific implementation. Since the PTT does not need to use these 16 bits, all 16 bits are set to zero.
  • the present invention adds an extended portion of the RTP fixed header to the extended bit position 1 of the RTP fixed header in the RTP voice data packet to be transmitted.
  • the format of the PTT header extension is:
  • the first 16 bits are all set to 0, and the length is 1, indicating that only the data source identifier of the current call is added, and the data source identifier of the current call is 4 bytes. Indicates the data source identifier that currently holds the floor, that is, the PTT client that holds the floor. If no PTT client gets the floor, the data source ID of the current call is all set to 0.
  • the extension of the RTP of the present invention is mainly for adding a data source identifier of a current call, the data source identifier of the current call is set by the PTT client, and is unique, and the PTT server can decide which PTT client has a speech according to the PTT server. right.
  • the PTT server After receiving the RTP voice data packet sent by the PTT client for the header extension, the PTT server reads the data source identifier of the current call in the extended portion of the RTP voice data packet, and then according to the value of the data source identifier of the current call.
  • the range of the interval determines the speaking authority of the PTT client.
  • the PTT client when the value of the data source identifier of the current call is 0x1, the PTT client has a preemption right; when the value of the data source identifier of the current call is in the interval 0x2-0xFFFFFF, the PTT client does not have a preemption right, and applies for obtaining a floor when no third-party PTT client speaks; when the value of the data source identifier of the current call is in the interval Ox 1000000-OxFFFFFFFF, The PTT client has no say.
  • the PTT server performs corresponding operations according to the speaking authority of the PTT client. If there is a floor, the RTP voice data packet sent by the PTT client is sent to the third-party PTT client in a multicast manner, and the third-party PTT client can listen to the call content of the PTT client through the RTP voice data packet, which is well implemented.
  • the group call function is well implemented.
  • the PTT server will judge the RTP voice data packet. Whether it contains voice data. If there is no voice data, it is determined that the PTT client gives up the current floor; and if there is voice data, the user level is distinguished by the data source identifier of the current call, thereby determining the floor.
  • step S14 is performed, otherwise step S13 is performed;
  • step S16 is performed;
  • step S18 is performed
  • step S20 is performed;
  • the invention performs header expansion on the RTP voice data packet sent by the PTT client, and sets the data source identifier of the current call in the header extension part, so that the PTT server decides the speaking permission of the PTT client according to the currently adopted data source identifier, and the speaker will have a speech.
  • the RTP voice data packet sent by the PTT client of the right is sent to the third-party PTT client in multicast form to implement the group call function.
  • the implementation of the header extension of the RTP voice data packet is relatively simple and easy to implement, and the PTT server can make a correct decision only through a data source identifier, which facilitates the design and promotion of the system.

Landscapes

  • Engineering & Computer Science (AREA)
  • Multimedia (AREA)
  • Computer Networks & Wireless Communication (AREA)
  • Signal Processing (AREA)
  • Telephonic Communication Services (AREA)

Abstract

Disclosed are an implementation method and implementation system for a PTT call based on VoIP technology. A header extension is performed on an RTP voice data packet transmitted to a PTT client, a data source identifier of a current call is configured in a header extension part, and a PTT server is allowed to determine a permission-to-speak of the PTT client on the basis of the data source identifier currently being passed through, and to transmit to a third-party PTT client in the form of a multicast the RTP voice data packet transmitted by the PTT client having the permission-to-speak, thus implementing a group call function. Also, the implementation of the performance of the header extension on the RTP voice data packet is relatively simple and easy to implement, and only one data source identifier is needed to allow the PTT server to make a correct determination, thus facilitating system design and promotion.

Description

一种基于 VOIP技术的 PTT通话实现方法及实现系统  PTT call implementation method and implementation system based on VOIP technology
【技术领域】 [Technical Field]
本发明涉及移动终端通信技术的应用领域, 尤其涉及的是一种基于 VOIP 技术的 PTT通话实现方法及实现系统。  The invention relates to the application field of mobile terminal communication technology, in particular to a PTT call implementation method and implementation system based on VOIP technology.
【背景技术】 【Background technique】
VoIP的全称为 Voice On IP, 即为基于 Internet的通话, 从使用的角度看, 它类似于通常的手机能实现点对点的通话。 VoIP最大的优点就是通话便宜, 但 不能实现一对多的通话。  The full name of VoIP is Voice On IP, which is an Internet-based call. From the perspective of usage, it can achieve peer-to-peer calls like a normal mobile phone. The biggest advantage of VoIP is that the call is cheap, but one-to-many calls cannot be achieved.
PTT的全称为 Push To Talk,即一键通话,其功能类似于传统的对讲机。 PTT 的优点是具有群呼功能。 它可以设置成组, 比如说有十个人设置成一组, 那么 一个人在讲话, 其他九个人都能够听到。 公司、 机构(如酒店、 机场)或组织(如 施工队)的工作人员, 都可以形成一个组来使用这种群呼功能, 而不必要一个一 个地打电话通知。  The full name of PTT is Push To Talk, which is a one-touch call, which functions like a traditional walkie-talkie. The advantage of PTT is that it has a group call function. It can be grouped into groups, for example, if there are ten people set up in a group, then one person is speaking and the other nine can hear it. Companies, organizations (such as hotels, airports) or organizations (such as construction teams) can form a group to use this group call function, and do not have to call one by one.
随着通信技术的发展, 基于 VOIP技术实现 PTT群呼通话成为了可能, 但 现有实现方法和系统多过于复杂, 不利于推广。  With the development of communication technology, it is possible to implement PTT group call based on VOIP technology, but the existing implementation methods and systems are too complicated and not conducive to promotion.
因此, 现有技术还有待于改进和发展。  Therefore, the prior art has yet to be improved and developed.
【发明内容】 [Summary of the Invention]
本发明要解决的技术问题在于, 针对现有技术的上述缺陷, 提供一种基于 The technical problem to be solved by the present invention is to provide a basis for the above-mentioned drawbacks of the prior art.
VOIP技术的 PTT通话实现方法及实现系统, 解决 VOIP的群呼功能, 并使得基 于 VOIP技术实现 PTT群呼通话的设计简单、 易推广。 The PTT call implementation method and implementation system of VOIP technology solves the group call function of VOIP, and makes the design of PTT group call based on VOIP technology simple and easy to promote.
为解决上述技术问题, 本发明釆用的一个技术方案是提供一种基于 VOIP 技术的 PTT通话实现方法, 包括以下步骤: A、 通过具有 VOIP功能的 PTT客 户端对待发送的 RTP语音数据包进行头扩展, 在头扩展部分设置一当前通话的 数据源标识, 并将头扩展后的 RTP语音数据包发送给具有 VOIP功能的 PTT服 务器; 其中, 所述步骤 A包括: 所述 PTT客户端将待发送的 RTP语音数据包中 RTP固定头的扩展比特位置 1 , 为 RTP固定头增加所述头扩展部分; B、 所述 PTT服务器读取所述头扩展后的 RTP语音数据包中的当前通话的数据源标识, 并根据所述当前通话的数据源标识的值所处的区间范围判断所述 PTT客户端的 发言权限; 其中, 所述步骤 B还包括: 当所述当前通话的数据源标识的值为 0x1 时, 所述 PTT客户端具有发言抢占权; 当所述当前通话的数据源标识的值处于 区间 0x2-0xFFFFFF时, 所述 PTT客户端没有发言抢占权, 且在无第三方 PTT 客户端发言时申请获得发言权; 当所述当前通话的数据源标识的值处于区间 Ox 1000000-OxFFFFFFFF时, 所述 PTT客户端没有发言权; C、 当判断所述 PTT 客户端有发言权, 所述 PTT服务器将所述头扩展后的 RTP语音数据包以组播形 式发送给第三方 PTT客户端。 In order to solve the above technical problem, one technical solution adopted by the present invention is to provide a method for implementing a PTT call based on VOIP technology, including the following steps: A. Heading a RTP voice data packet to be sent by a PTT client having a VOIP function Expanding, setting a data source identifier of the current call in the header extension, and transmitting the header-expanded RTP voice data packet to the PTIP-enabled PTT server; wherein the step A includes: the PTT client is to be sent The extended bit position 1 of the RTP fixed header in the RTP voice data packet increases the header extension portion for the RTP fixed header; B. The PTT server reads the data of the current call in the RTP voice data packet after the header extension Source identifier, and determining, according to the range of the interval in which the value of the data source identifier of the current call is located, determining the PTT client The step B further includes: when the value of the data source identifier of the current call is 0x1, the PTT client has a preemption right; when the value of the data source identifier of the current call is in the interval 0x2-0xFFFFFF, the PTT client does not have a preemption right, and applies for obtaining a floor when no third party PTT client speaks; when the value of the data source identifier of the current call is in the interval Ox 1000000-OxFFFFFFFF, The PTT client has no floor. C. When it is determined that the PTT client has a floor, the PTT server sends the extended RTP voice packet to the third-party PTT client in multicast form.
其中, 所述步骤 B还包括: 所述 PTT服务器接收到所述头扩展后的 RTP语 音数据包时, 判断所述头扩展后的 RTP语音数据包中是否有语音数据; 当所述 头扩展后的 RTP语音数据包中没有语音数据时判定该 PTT客户端放弃发言权, 否则通过所述当前通话的数据源标识的值所处的区间范围判断所述 PTT客户端 的发言权限。  The step B further includes: when the PTT server receives the extended RTP voice data packet, determining whether there is voice data in the extended RTP voice data packet; when the header is expanded When there is no voice data in the RTP voice data packet, it is determined that the PTT client gives up the floor, otherwise the speaking permission of the PTT client is determined by the range of the range in which the value of the data source identifier of the current call is located.
其中, 所述步骤 A中的当前通话的数据源标识唯一标识一 PTT客户端, 用 于所述 PTT服务器据此判断所述 PTT客户端的发言权限。  The data source identifier of the current call in the step A uniquely identifies a PTT client, and the PTT server determines the speaking permission of the PTT client accordingly.
其中, 所述当前通话的数据源标识长度为 4个字节。  The data source identifier of the current call is 4 bytes in length.
为解决上述技术问题, 本发明釆用的另一个技术方案是提供一种基于 VOIP 技术的 PTT通话实现方法, 其中, 包括以下步骤: A、 通过具有 VOIP功能的 PTT客户端对待发送的 RTP语音数据包进行头扩展, 在头扩展部分设置一当前 通话的数据源标识, 并将头扩展后的 RTP语音数据包发送给具有 VOIP功能的 PTT服务器; B、 所述 PTT服务器读取所述头扩展后的 RTP语音数据包中的当 前通话的数据源标识, 并根据所述当前通话的数据源标识的值所处的区间范围 判断所述 PTT客户端的发言权限; C、 当判断所述 PTT客户端有发言权, 所述 PTT服务器将所述头扩展后的 RTP语音数据包以组播形式发送给第三方 PTT客 户端。  In order to solve the above technical problem, another technical solution adopted by the present invention is to provide a method for implementing a PTT call based on VOIP technology, which includes the following steps: A. RTP voice data to be sent by a PTT client having a VOIP function The packet is extended by the header, the data source identifier of the current call is set in the header extension part, and the extended RTP voice data packet is sent to the POT server with the VOIP function; B. After the PTT server reads the header extension The data source identifier of the current call in the RTP voice data packet, and determining the speaking permission of the PTT client according to the range of the range in which the value of the data source identifier of the current call is located; C. determining that the PTT client has The floor, the PTT server sends the extended RTP voice data packet to the third-party PTT client in multicast form.
其中, 所述步骤 A包括: 所述 PTT客户端将待发送的 RTP语音数据包中 RTP固定头的扩展比特位置 1 , 为 RTP固定头增加所述头扩展部分。  The step A includes: the PTT client adds the extended bit position 1 of the RTP fixed header in the RTP voice data packet to be sent, and adds the header extension to the RTP fixed header.
其中, 所述步骤 B还包括: 当所述当前通话的数据源标识的值为 0x1时, 所述 PTT客户端具有发言抢占权; 当所述当前通话的数据源标识的值处于区间 0x2-0xFFFFFF时, 所述 PTT客户端没有发言抢占权, 且在无第三方 PTT客户 端发言时申请获得发言权; 当所述当前通话的数据源标识的值处于区间 Ox 1000000-OxFFFFFFFF时, 所述 PTT客户端没有发言权。 θ. 其中, 所述步骤 B还包括: 所述 PTT服务器接收到所述头扩展后的 RTP语 音数据包时, 判断所述头扩展后的 RTP语音数据包中是否有语音数据; 当所述 头扩展后的 RTP语音数据包中没有语音数据时判定该 PTT客户端放弃发言权, 否则通过所述当前通话的数据源标识的值所处的区间范围判断所述 PTT客户端 的发言权限。 The step B further includes: when the value of the data source identifier of the current call is 0x1, the PTT client has a preemption right; when the value of the data source identifier of the current call is in the interval 0x2-0xFFFFFF The PTT client does not have the right to speak, and applies for obtaining the floor when no third-party PTT client speaks; when the value of the data source identifier of the current call is in the interval Ox 1000000-OxFFFFFFFF, the PTT client There is no right to speak. θ. The step B further includes: when the PTT server receives the extended RTP voice data packet, determining whether there is voice data in the extended RTP voice data packet; When there is no voice data in the extended RTP voice data packet, it is determined that the PTT client gives up the floor, otherwise the speaking permission of the PTT client is determined by the range of the range in which the value of the data source identifier of the current call is located.
其中, 所述步骤 A中的当前通话的数据源标识唯一标识一 PTT客户端, 用 于所述 PTT服务器据此判断所述 PTT客户端的发言权限。  The data source identifier of the current call in the step A uniquely identifies a PTT client, and the PTT server determines the speaking permission of the PTT client accordingly.
其中, 所述当前通话的数据源标识长度为 4个字节。  The data source identifier of the current call is 4 bytes in length.
为解决上述技术问题, 本发明采用的再一个技术方案是提供一种基于 VOIP 技术的 PTT通话实现系统, 其中, 包括具有 VOIP功能的 PTT客户端和 PTT服 务器, 其中, 所述 PTT客户端用于对 RTP语音数据包进行头扩展, 在头扩展部 分设置一当前通话的数据源标识, 并将头扩展后的 RTP语音数据包发送给所述 PTT服务器; 所述 PTT服务器用于接收所述 PTT客户端发送的头扩展后的 RTP 语音数据包, 并获取所述当前通话的数据源标识, 通过所述当前通话的数据源 标识判断所述 PTT客户端的发言权限, 并根据所述 PTT客户端的发言权限做出 是否将 RTP语音数据包以组播的方式发送给第三方 PTT客户端的操作。  In order to solve the above technical problem, another technical solution adopted by the present invention is to provide a PTT call implementation system based on VOIP technology, which includes a PTT client and a PTT server having a VOIP function, wherein the PTT client is used for Performing header expansion on the RTP voice data packet, setting a data source identifier of the current call in the header extension portion, and transmitting the header extended RTP voice data packet to the PTT server; the PTT server is configured to receive the PTT client Sending the extended RTP voice data packet, and obtaining the data source identifier of the current call, determining the speaking permission of the PTT client by using the data source identifier of the current call, and according to the speaking permission of the PTT client Make an operation of whether to send the RTP voice data packet to the third-party PTT client in a multicast manner.
其中, 所述 PTT客户端和 PTT服务器均为具有 VOIP功能的移动终端, 且 所述 PTT客户端为多个, 所述 PTT服务器为一个。  The PTT client and the PTT server are both VOIP-enabled mobile terminals, and the PTT client is multiple, and the PTT server is one.
本发明所提供的基于 VOIP技术的 PTT通话实现方法及实现系统, 通过对 PTT客户端发送的 RTP语音数据包进行头扩展, 在头扩展部分设置当前通话的 数据源标识, 以便 PTT服务器根据当前通过的数据源标识裁决出 PTT客户端的 发言权限, 将有发言权的 PTT客户端发送的 RTP语音数据包以组播形式发送给 第三方 PTT客户端, 实现群呼功能。 而且, 对 RTP语音数据包进行头扩展的实 现方式比较筒单, 易于实现, 仅通过一数据源标识即可使 PTT服务器做出正确 的裁决, 便于系统的设计和推广。  The method and the implementation system for implementing the PTT call based on the VOIP technology provided by the present invention, by performing header expansion on the RTP voice data packet sent by the PTT client, setting the data source identifier of the current call in the header extension portion, so that the PTT server can pass the current pass. The data source identifier determines the speaking permission of the PTT client, and the RTP voice data packet sent by the PTT client with the floor is sent to the third-party PTT client in multicast form to implement the group call function. Moreover, the implementation of the header extension of the RTP voice data packet is relatively simple, and is easy to implement. The PTT server can make a correct decision only through a data source identifier, which facilitates the design and promotion of the system.
【附图说明】  [Description of the Drawings]
图 1是本发明提供的基于 VOIP技术的 PTT通话实现系统的结构示意图。 图 2是本发明提供的基于 VOIP技术的 PTT通话实现方法的流程图。  FIG. 1 is a schematic structural diagram of a PTT call implementation system based on VOIP technology provided by the present invention. 2 is a flow chart of a method for implementing a PTT call based on VOIP technology provided by the present invention.
图 3是本发明提供的基于 VOIP技术的 PTT通话实现方法中 PTT服务器对 PTT客户端的发言权限进行裁决的方法流程图。  FIG. 3 is a flowchart of a method for determining a speaking right of a PTT client by a PTT server according to a VOIP-based PTT call implementation method provided by the present invention.
图 4是本发明提供的基于 VOIP技术的 PTT通话实现方法中 PTT服务器进 行裁决的方法流程图。  4 is a flow chart of a method for determining a PTT server in a PTT call implementation method based on VOIP technology provided by the present invention.
更正页 (½则第 9 1 条) 【具体实施方式】 Correction page (1⁄2 then Article 9 1) 【detailed description】
为使本发明的目的、 技术方案及优点更加清楚、 明确, 以下参照附图并举 实施例对本发明进一步详细说明。 应当理解, 此处所描述的具体实施例仅仅用 以解释本发明, 并不用于限定本发明。  The present invention will be further described in detail below with reference to the accompanying drawings. It is understood that the specific embodiments described herein are merely illustrative of the invention and are not intended to limit the invention.
本发明提供一种基于 VOIP技术的 PTT通话实现方法及实现系统, 如图 1 所示, 所述系统包括数个 PTT客户端和一个 PTT服务器, PTT客户端和 PTT 服务器均为具有 VOIP (基于互联网的通话)功能的移动终端, PTT客户端可以 是用户使用的手机, 用于用户进行 PTT通话, 而为了提高数据处理速率, PTT 服务器可以为电脑, 用于语音接收与组播、 以及裁决哪个 PTT客户端有权说话。  The present invention provides a method and an implementation system for implementing a PTT call based on VOIP technology. As shown in FIG. 1, the system includes a plurality of PTT clients and a PTT server, and both the PTT client and the PTT server have VOIP (based on the Internet). The mobile terminal of the function of the call, the PTT client can be a mobile phone used by the user for the user to make a PTT call, and in order to improve the data processing rate, the PTT server can be used for the computer, for voice receiving and multicasting, and which PTT is decided. The client has the right to speak.
该系统通过软件实现, 每个 PTT客户端通过 RTP (实时传输协议 )与 PTT 服务器进行数据传输, PTT服务器对 PTT客户端发送的 RTP语音数据包进行裁 决, 根据 RTP语音数据包中的数据源标识判断 PTT客户端的发言权限, 并根据 PTT客户端的发言权限, 以组播形式向第三方 PTT客户端发布该 RTP语音数据 包。  The system is implemented by software. Each PTT client performs data transmission with the PTT server through RTP (Real-Time Transport Protocol), and the PTT server arbitrates the RTP voice data packet sent by the PTT client according to the data source identifier in the RTP voice data packet. The PTT client's speaking permission is determined, and the RTP voice data packet is distributed to the third-party PTT client in a multicast form according to the speaking permission of the PTT client.
具体地, 所述 PTT客户端用于对 RTP语音数据包进行头扩展,在 RTP语音 数据包的 RTP固定头后增加一头扩展部分, 并在头扩展部分设置一当前通话的 数据源标识,然后将头扩展后的 RTP语音数据包发送给所述 PTT服务器;其中, 该当前通话的数据源标识用于标识 PTT客户端当前通话的发言权限, 通过该当 前通话的数据源标识的值, PTT服务器可以判断出 PTT客户端的发言权限。  Specifically, the PTT client is used for header expansion of the RTP voice data packet, adding an extension part after the RTP fixed header of the RTP voice data packet, and setting a data source identifier of the current call in the header extension part, and then The extended RTP voice data packet is sent to the PTT server, where the data source identifier of the current call is used to identify the speaking permission of the current call of the PTT client, and the PTT server may use the value of the data source identifier of the current call. Determine the speaking permission of the PTT client.
所述 PTT服务器用于接收所述 PTT客户端发送的头扩展后的 RTP语音数据 包, 并获取所述当前通话的数据源标识, 通过所述当前通话的数据源标识判断 所述 PTT客户端的发言权限, 并根据所述 PTT客户端的发言权限做出是否将 RTP语音数据包以组播的方式发送给第三方 PTT客户端的操作。  The PTT server is configured to receive a header extended RTP voice data packet sent by the PTT client, obtain a data source identifier of the current call, and determine, by using a data source identifier of the current call, a speech of the PTT client. Authorization, and according to the speaking permission of the PTT client, whether to send the RTP voice data packet to the third-party PTT client in a multicast manner.
结合上述本发明提供的基于 VOIP技术的 PTT通话实现系统, 本发明还提 供了一种基于 VOIP技术的 PTT通话实现方法, 如图 2所示, 包括以下步骤: 步骤 S100、通过具有 VOIP功能的 PTT客户端对待发送的 RTP语音数据包 进行头扩展,在头扩展部分设置一当前通话的数据源标识,并将头扩展后的 RTP 语音数据包发送给具有 VOIP功能的 PTT服务器;  In combination with the VOIP-based PTT call implementation system provided by the present invention, the present invention further provides a PTIP-based PTT call implementation method. As shown in FIG. 2, the method includes the following steps: Step S100: Passing a POT with VOIP function The client performs header expansion on the RTP voice data packet to be sent, sets a data source identifier of the current call in the header extension portion, and sends the extended RTP voice data packet to the POT server with the VOIP function;
步骤 S200、所述 PTT服务器读取所述头扩展后的 RTP语音数据包中的当前 通话的数据源标识, 并根据所述当前通话的数据源标识的值所处的区间范围判 断所述 PTT客户端的发言权限; Step S200: The PTT server reads the data source identifier of the current call in the extended RTP voice data packet, and judges according to the range of the value of the data source identifier of the current call. Disconnecting the speaking permission of the PTT client;
步骤 S300、 当判断所述 PTT客户端有发言权,所述 PTT服务器将所述头扩 展后的 RTP语音数据包以组播形式发送给第三方 PTT客户端。  Step S300: When it is determined that the PTT client has a floor, the PTT server sends the extended RTP voice data packet to a third-party PTT client in multicast form.
下面结合具体的实施例对上述步骤进行详细的说明。  The above steps will be described in detail below with reference to specific embodiments.
在实际通话过程中,多个 PTT客户端会同时通过 RTP协议向 PTT服务器发 送 RTP语音数据包, PTT服务器会对接收到的各个 RTP语音数据包进行裁决, 裁决哪个 PTT客户端具有发言权,并将有发言权的 PTT客户端的 RTP语音数据 包以组播形式发送给第三方 PTT客户端, 从而实现群呼功能。  During the actual call, multiple PTT clients will simultaneously send RTP voice packets to the PTT server through the RTP protocol, and the PTT server will decide on each RTP voice packet received, and decide which PTT client has the floor, and The RTP voice data packet of the PTT client with the voice is sent to the third-party PTT client in multicast form, thereby implementing the group call function.
为了便于 PTT服务器对 PTT客户端做出正确的裁决, PTT客户端对待发送 的 RTP语音数据包进行头扩展, RTP固定头的格式如下:  In order to facilitate the PTT server to make a correct ruling on the PTT client, the PTT client performs header extension on the RTP voice packet to be sent. The format of the RTP fixed header is as follows:
01234567890123456789012345678901  01234567890123456789012345678901
+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_  +_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_ +_+_+_+_+_+_+_
|V=2|P|X| CC |M| PT | 序列号 |  |V=2|P|X| CC |M| PT | Serial Number |
+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_  +_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_ +_+_+_+_+_+_+_
I 时间戳 I I time stamp I
_+_+_+_++_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+__+_+_+_++_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_ +_+_+_+_+_+_+_+_+_+_+_
1 数据源标识 I1 Data source identification I
+ =+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=++ =+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+= +=+=+=+
I 贡 献数据 源 标识 II Contribution Data Source I
I …- I I ...- I
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+--+-+-+-+ 其中, 数据源标识用于标识不同的数据源, 每个数据源都有一个数据源标 识, 并且有数据源设定。 X表示扩展比特位, 如果 X置 1, 则表示固定头后有 一头扩展部分。 而头扩展部分的格式为:  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+- +-+-+-+-+-+-+--+-+-+-+ where the data source identifier is used to identify different data sources, each data source has a data source identifier, and there is a data source set up. X represents the extended bit. If X is set to 1, it means that there is an extended part after the fixed header. The format of the header extension is:
01234567890123456789012345678901  01234567890123456789012345678901
+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_  +_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_ +_+_+_+_+_+_+_
1 定义格式 I length | +_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_  1 Define the format I length | +_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_ +_+_+_+_+_+_+_+_+_+_
I 头扩展 I I header extension I
I …- II ...- I
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+—+- 其中, 头扩展部分包括 16比特的长度域, 指示扩展项中 32比特字的个数。 RTP固定头之后只允许有一个头扩展.扩展项的前 16比特用以识别标识符或参数. 这 16比特的格式不在 RTP中说明, 而是由具体实现的上层协议定义。 由于 PTT 并不需要用到这 16位, 因此将这 16位全部置 0。 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+- +-+-+-+-+-+-+- where the header extension includes a 16-bit length field indicating the number of 32-bit words in the extension. Only one header extension is allowed after the RTP fixed header. The first 16 bits of the extension are used to identify the identifier or parameter. This 16-bit format is not specified in the RTP, but is defined by the upper layer protocol of the specific implementation. Since the PTT does not need to use these 16 bits, all 16 bits are set to zero.
因此,本发明将待发送的 RTP语音数据包中 RTP固定头的扩展比特位置 1, 为 RTP固定头增加一头扩展部分。 其中, PTT头扩展部分的格式为:  Therefore, the present invention adds an extended portion of the RTP fixed header to the extended bit position 1 of the RTP fixed header in the RTP voice data packet to be transmitted. The format of the PTT header extension is:
01234567890123456789012345678901  01234567890123456789012345678901
+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_  +_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_ +_+_+_+_+_+_+_
+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_ +_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_+_ +_+_+_+_+_+_+_
1 当前通话的数据源标识 I  1 Data source identifier of the current call I
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+- 其中,前 16位全部置 0, length为 1,表示只增加了当前通话的数据源标识, 而当前通话的数据源标识为 4个字节, 表示当前持有发言权的数据源标识, 也 即是标识持有发言权的 PTT客户端。 如果没有 PTT客户端得到发言权, 则当前 通话的数据源标识全部置 0。  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+- +-+-+-+-+-+-+- where the first 16 bits are all set to 0, and the length is 1, indicating that only the data source identifier of the current call is added, and the data source identifier of the current call is 4 bytes. Indicates the data source identifier that currently holds the floor, that is, the PTT client that holds the floor. If no PTT client gets the floor, the data source ID of the current call is all set to 0.
本发明对 RTP的扩展, 主要就是为了增加一个当前通话的数据源标识, 该 当前通话的数据源标识由 PTT客户端设定, 并且是唯一的, PTT服务器可以据 此裁决哪个 PTT客户端有发言权。  The extension of the RTP of the present invention is mainly for adding a data source identifier of a current call, the data source identifier of the current call is set by the PTT client, and is unique, and the PTT server can decide which PTT client has a speech according to the PTT server. right.
在 PTT服务器接收到 PTT客户端发送的进行头扩展后的 RTP语音数据包 时, 读取 RTP语音数据包中头扩展部分的当前通话的数据源标识, 然后根据当 前通话的数据源标识的值所处的区间范围判断所述 PTT客户端的发言权限。  After receiving the RTP voice data packet sent by the PTT client for the header extension, the PTT server reads the data source identifier of the current call in the extended portion of the RTP voice data packet, and then according to the value of the data source identifier of the current call. The range of the interval determines the speaking authority of the PTT client.
从图 3所示的表中可以看出, 当所述当前通话的数据源标识的值为 0x1时, 所述 PTT客户端具有发言抢占权; 当所述当前通话的数据源标识的值处于区间 0x2-0xFFFFFF时, 所述 PTT客户端没有发言抢占权, 且在无第三方 PTT客户 端发言时申请获得发言权; 当所述当前通话的数据源标识的值处于区间 Ox 1000000-OxFFFFFFFF时, 所述 PTT客户端没有发言权。  As can be seen from the table shown in FIG. 3, when the value of the data source identifier of the current call is 0x1, the PTT client has a preemption right; when the value of the data source identifier of the current call is in the interval 0x2-0xFFFFFF, the PTT client does not have a preemption right, and applies for obtaining a floor when no third-party PTT client speaks; when the value of the data source identifier of the current call is in the interval Ox 1000000-OxFFFFFFFF, The PTT client has no say.
PTT服务器根据 PTT客户端的发言权限, 进行相应的操作。 如果有发言权 则将 PTT客户端发送的 RTP语音数据包以组播形式发送给第三方 PTT客户端, 第三方 PTT客户端可以通过 RTP语音数据包收听该 PTT客户端的通话内容,很 好的实现了群呼功能。。  The PTT server performs corresponding operations according to the speaking authority of the PTT client. If there is a floor, the RTP voice data packet sent by the PTT client is sent to the third-party PTT client in a multicast manner, and the third-party PTT client can listen to the call content of the PTT client through the RTP voice data packet, which is well implemented. The group call function. .
当然, PTT服务器在接收到 RTP语音数据包后, 会判断 RTP语音数据包中 是否包含有语音数据。 如果没有语音数据, 则判定该 PTT客户端放弃当前的发 言权; 而如果有语音数据, 则通过当前通话的数据源标识来区分用户级别, 从 而进行发言权的裁决。 Of course, after receiving the RTP voice data packet, the PTT server will judge the RTP voice data packet. Whether it contains voice data. If there is no voice data, it is determined that the PTT client gives up the current floor; and if there is voice data, the user level is distinguished by the data source identifier of the current call, thereby determining the floor.
如图 4所示, PTT服务器进行裁决的具体步骤如下:  As shown in Figure 4, the specific steps for the PTT server to make a ruling are as follows:
511、 是否接收到 RTP语音数据包; 如果是, 执行步骤 S12, 否则继续判断 是否接收到 RTP语音数据包;  511. Whether the RTP voice data packet is received; if yes, go to step S12, otherwise continue to determine whether the RTP voice data packet is received;
512、 RTP语音数据包中是否有语音数据; 如果是, 执行步骤 S14, 否则执 行步骤 S13 ;  512, whether there is voice data in the RTP voice data packet; if yes, step S14 is performed, otherwise step S13 is performed;
513、 放弃发言权, 并返回步骤 S11 ;  513, give up the right to speak, and return to step S11;
514、 根据当前通话的数据源标识的值, 判断 PTT客户端的发言权限; 514. Determine, according to the value of the data source identifier of the current call, the speaking permission of the PTT client.
515、如果当前通话的数据源标识的值为 0x1 ,则 PTT客户端为超级发言者, 并执行步骤 S16; 515, if the value of the data source identifier of the current call is 0x1, the PTT client is a super speaker, and step S16 is performed;
516、 获得发言权, 并返回步骤 S11 ;  516, obtain the right to speak, and return to step S11;
517、 如果当前通话的数据源标识的值处于区间 0x2-0xFFFFFF, 则 PTT客 户端为发言者, 并执行步骤 S18;  517. If the value of the data source identifier of the current call is in the interval 0x2-0xFFFFFF, the PTT client is the speaker, and step S18 is performed;
518、 当前是否有 PTT客户端在发言, 如果是执行步骤 S20, 否则执行步骤 518. Is there currently a PTT client speaking, if it is performing step S20, otherwise executing steps
S16; S16;
519、 如果当前通话的数据源标识的值处于区间 OxlOOOOOO-OxFFFFFFFF , 则 PTT客户端为听者, 并执行步骤 S20;  519, if the value of the data source identifier of the current call is in the interval OxlOOOOOO-OxFFFFFFFF, the PTT client is the listener, and step S20 is performed;
520、 没有发言权, 并返回步骤 Sll。  520, no speaking, and return to step S11.
本发明通过对 PTT客户端发送的 RTP语音数据包进行头扩展, 在头扩展部 分设置当前通话的数据源标识, 以便 PTT服务器根据当前通过的数据源标识裁 决出 PTT客户端的发言权限,将有发言权的 PTT客户端发送的 RTP语音数据包 以组播形式发送给第三方 PTT客户端, 实现群呼功能。 而且, 对 RTP语音数据 包进行头扩展的实现方式比较简单,易于实现,仅通过一数据源标识即可使 PTT 服务器做出正确的裁决, 便于系统的设计和推广。  The invention performs header expansion on the RTP voice data packet sent by the PTT client, and sets the data source identifier of the current call in the header extension part, so that the PTT server decides the speaking permission of the PTT client according to the currently adopted data source identifier, and the speaker will have a speech. The RTP voice data packet sent by the PTT client of the right is sent to the third-party PTT client in multicast form to implement the group call function. Moreover, the implementation of the header extension of the RTP voice data packet is relatively simple and easy to implement, and the PTT server can make a correct decision only through a data source identifier, which facilitates the design and promotion of the system.
应当理解的是, 本发明的应用不限于上述的举例, 对本领域普通技术人员 来说, 可以根据上述说明加以改进或变换, 所有这些改进和变换都应属于本发 明所附权利要求的保护范围。  It is to be understood that the application of the present invention is not limited to the above-described examples, and those skilled in the art can make modifications and changes in accordance with the above description, all of which are within the scope of the appended claims.

Claims

权利 要求 Rights request
1. 一种基于 VOIP技术的 PTT通话实现方法,其特征在于, 包括以下步骤:A method for implementing a PTT call based on VOIP technology, comprising the steps of:
A、 通过具有 VOIP功能的 PTT客户端对待发送的 RTP语音数据包进行头 扩展, 在头扩展部分设置一当前通话的数据源标识, 并将头扩展后的 RTP语音 数据包发送给具有 VOIP功能的 PTT服务器;其中,所述步骤 A包括:所述 PTT 客户端将待发送的 RTP语音数据包中 RTP固定头的扩展比特位置 1 , 为 RTP固 定头增加所述头扩展部分; A. The header of the RTP voice data packet to be sent by the PTT client with the VOIP function is extended, the data source identifier of the current call is set in the header extension part, and the extended RTP voice data packet is sent to the VOIP-enabled function. a PTT server, wherein the step A includes: the PTT client adds an extended bit position 1 of the RTP fixed header in the RTP voice data packet to be sent, and adds the header extension portion to the RTP fixed header;
B、 所述 PTT服务器读取所述头扩展后的 RTP语音数据包中的当前通话的 数据源标识, 并根据所述当前通话的数据源标识的值所处的区间范围判断所述 PTT客户端的发言权限; 其中, 所述步骤 B还包括:  B. The PTT server reads the data source identifier of the current call in the extended RTP voice data packet, and determines the PTT client according to the range of the range in which the value of the data source identifier of the current call is located. The speaking permission; wherein, the step B further includes:
当所述当前通话的数据源标识的值为 0x1时, 所述 PTT客户端具有发言抢 占权;  When the value of the data source identifier of the current call is 0x1, the PTT client has a preemption right;
当所述当前通话的数据源标识的值处于区间 0x2-0xFFFFFF时, 所述 PTT 客户端没有发言抢占权, 且在无第三方 PTT客户端发言时申请获得发言权; 当所述当前通话的数据源标识的值处于区间 OxlOOOOOO-OxFFFFFFFF时,所 述 PTT客户端没有发言权;  When the value of the data source identifier of the current call is in the interval 0x2-0xFFFFFF, the PTT client does not have the preemption right, and applies for obtaining the floor when no third party PTT client speaks; when the current call data When the value of the source identifier is in the interval Ox1000OOO-OxFFFFFFFF, the PTT client has no floor;
C、 当判断所述 PTT客户端有发言权, 所述 PTT服务器将所述头扩展后的 RTP语音数据包以组播形式发送给第三方 PTT客户端。  C. When it is determined that the PTT client has a floor, the PTT server sends the extended RTP voice packet to the third-party PTT client in multicast form.
2. 根据权利要求 1所述的基于 VOIP技术的 PTT通话实现方法, 其特征在 于, 所述步骤 B还包括:  The VOIP-based PTT call implementation method according to claim 1, wherein the step B further includes:
所述 PTT服务器接收到所述头扩展后的 RTP语音数据包时, 判断所述头扩 展后的 RTP语音数据包中是否有语音数据;  And receiving, by the PTT server, the RTP voice data packet after the header extension, determining whether there is voice data in the extended RTP voice data packet;
当所述头扩展后的 RTP语音数据包中没有语音数据时判定该 PTT客户端放 弃发言权, 否则通过所述当前通话的数据源标识的值所处的区间范围判断所述 PTT客户端的发言权限。  Determining that the PTT client gives up the floor when there is no voice data in the extended RTP voice data packet, otherwise determining the speaking permission of the PTT client by the range of the range in which the value of the data source identifier of the current call is located .
3. 根据权利要求 1所述的基于 VOIP技术的 PTT通话实现方法, 其特征在 于, 所述步骤 A中的当前通话的数据源标识唯一标识一 PTT客户端, 用于所述 PTT服务器据此判断所述 PTT客户端的发言权限。  The VOIP-based PTT call implementation method according to claim 1, wherein the data source identifier of the current call in the step A uniquely identifies a PTT client, and the PTT server determines The speaking permission of the PTT client.
4. 根据权利要求 3所述的基于 VOIP技术的 PTT通话实现方法, 其特征在 于, 所述当前通话的数据源标识长度为 4个字节。 The VOIP-based PTT call implementation method according to claim 3, wherein the data source identifier length of the current call is 4 bytes.
5. 一种基于 VOIP技术的 PTT通话实现方法,其特征在于, 包括以下步骤:A method for implementing a PTT call based on VOIP technology, comprising the steps of:
A、 通过具有 VOIP功能的 PTT客户端对待发送的 RTP语音数据包进行头 扩展, 在头扩展部分设置一当前通话的数据源标识, 并将头扩展后的 RTP语音 数据包发送给具有 VOIP功能的 PTT服务器; A. The header of the RTP voice data packet to be sent by the PTT client with the VOIP function is extended, the data source identifier of the current call is set in the header extension part, and the extended RTP voice data packet is sent to the VOIP-enabled function. PTT server;
B、 所述 PTT服务器读取所述头扩展后的 RTP语音数据包中的当前通话的 数据源标识, 并根据所述当前通话的数据源标识的值所处的区间范围判断所述 PTT客户端的发言权限;  B. The PTT server reads the data source identifier of the current call in the extended RTP voice data packet, and determines the PTT client according to the range of the range in which the value of the data source identifier of the current call is located. Right to speak;
C、 当判断所述 PTT客户端有发言权, 所述 PTT服务器将所述头扩展后的 RTP语音数据包以组播形式发送给第三方 PTT客户端。  C. When it is determined that the PTT client has a floor, the PTT server sends the extended RTP voice packet to the third-party PTT client in multicast form.
6. 根据权利要求 5所述的基于 VOIP技术的 PTT通话实现方法, 其特征在 于, 所述步骤 A包括:  The VOIP-based PTT call implementation method according to claim 5, wherein the step A includes:
所述 PTT客户端将待发送的 RTP语音数据包中 RTP固定头的扩展比特位置 1 , 为 RTP固定头增加所述头扩展部分。  The PTT client adds the extended bit position 1 of the RTP fixed header in the RTP voice data packet to be sent, and adds the header extended portion to the RTP fixed header.
7. 根据权利要求 5所述的基于 VOIP技术的 PTT通话实现方法, 其特征在 于, 所述步骤 B还包括:  The VOIP-based PTT call implementation method according to claim 5, wherein the step B further includes:
当所述当前通话的数据源标识的值为 0x1时, 所述 PTT客户端具有发言抢 占权;  When the value of the data source identifier of the current call is 0x1, the PTT client has a preemption right;
当所述当前通话的数据源标识的值处于区间 0x2-0xFFFFFF时, 所述 PTT 客户端没有发言抢占权, 且在无第三方 PTT客户端发言时申请获得发言权; 当所述当前通话的数据源标识的值处于区间 OxlOOOOOO-OxFFFFFFFF时,所 述 PTT客户端没有发言权。  When the value of the data source identifier of the current call is in the interval 0x2-0xFFFFFF, the PTT client does not have the preemption right, and applies for obtaining the floor when no third party PTT client speaks; when the current call data When the value of the source identifier is in the interval Ox1000OOO-OxFFFFFFFF, the PTT client has no floor.
8. 根据权利要求 5所述的基于 VOIP技术的 PTT通话实现方法, 其特征在 于, 所述步骤 B还包括:  The VOIP-based PTT call implementation method according to claim 5, wherein the step B further includes:
所述 PTT服务器接收到所述头扩展后的 RTP语音数据包时, 判断所述头扩 展后的 RTP语音数据包中是否有语音数据;  And receiving, by the PTT server, the RTP voice data packet after the header extension, determining whether there is voice data in the extended RTP voice data packet;
当所述头扩展后的 RTP语音数据包中没有语音数据时判定该 PTT客户端放 弃发言权, 否则通过所述当前通话的数据源标识的值所处的区间范围判断所述 PTT客户端的发言权限。  Determining that the PTT client gives up the floor when there is no voice data in the extended RTP voice data packet, otherwise determining the speaking permission of the PTT client by the range of the range in which the value of the data source identifier of the current call is located .
9. 根据权利要求 5所述的基于 VOIP技术的 PTT通话实现方法, 其特征在 于, 所述步骤 A中的当前通话的数据源标识唯一标识一 PTT客户端, 用于所述 PTT服务器据此判断所述 PTT客户端的发言权限。 The VOIP-based PTT call implementation method according to claim 5, wherein the data source identifier of the current call in the step A uniquely identifies a PTT client, and the PTT server determines The speaking permission of the PTT client.
10. 根据权利要求 9所述的基于 VOIP技术的 PTT通话实现方法, 其特征 在于, 所述当前通话的数据源标识长度为 4个字节。 The VOIP-based PTT call implementation method according to claim 9, wherein the data source identifier length of the current call is 4 bytes.
11.一种基于 VOIP技术的 PTT通话实现系统,其特征在于,包括具有 VOIP 功能的 PTT客户端和 PTT服务器, 其中,  A PTT call implementation system based on VOIP technology, comprising: a PTIP-enabled PTT client and a PTT server, wherein
所述 PTT客户端用于对 RTP语音数据包进行头扩展, 在头扩展部分设置一 当前通话的数据源标识, 并将头扩展后的 RTP语音数据包发送给所述 PTT服务 器;  The PTT client is configured to perform header expansion on the RTP voice data packet, set a data source identifier of the current call in the header extension portion, and send the extended RTP voice data packet to the PTT server.
所述 PTT服务器用于接收所述 PTT客户端发送的头扩展后的 RTP语音数据 包, 并获取所述当前通话的数据源标识, 通过所述当前通话的数据源标识判断 所述 PTT客户端的发言权限, 并根据所述 PTT客户端的发言权限做出是否将 RTP语音数据包以组播形式发送给第三方 PTT客户端的操作。  The PTT server is configured to receive a header extended RTP voice data packet sent by the PTT client, obtain a data source identifier of the current call, and determine, by using a data source identifier of the current call, a speech of the PTT client. Authorization, and according to the speaking permission of the PTT client, whether to send the RTP voice data packet to the third-party PTT client in a multicast manner.
12. 根据权利要求 11所述的基于 VOIP技术的 PTT通话实现系统, 其特征 在于, 所述 PTT客户端和 PTT服务器均为具有 VOIP功能的移动终端, 且所述 PTT客户端为多个, 所述 PTT服务器为一个。  The VOIP-based PTT call implementation system according to claim 11, wherein the PTT client and the PTT server are both VOIP-enabled mobile terminals, and the PTT client is multiple. The PTT server is one.
PCT/CN2013/077159 2012-09-17 2013-06-13 Implementation method and implementation system for ptt call based on voip technology WO2014040433A1 (en)

Applications Claiming Priority (2)

Application Number Priority Date Filing Date Title
CN201210343519.0 2012-09-17
CN2012103435190A CN102916939A (en) 2012-09-17 2012-09-17 Implementation method and implementation system of PTT (push-to-talk) call based on VOIP (voice over internet protocol) technology

Publications (1)

Publication Number Publication Date
WO2014040433A1 true WO2014040433A1 (en) 2014-03-20

Family

ID=47615173

Family Applications (1)

Application Number Title Priority Date Filing Date
PCT/CN2013/077159 WO2014040433A1 (en) 2012-09-17 2013-06-13 Implementation method and implementation system for ptt call based on voip technology

Country Status (2)

Country Link
CN (1) CN102916939A (en)
WO (1) WO2014040433A1 (en)

Cited By (1)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
WO2022073219A1 (en) * 2020-10-10 2022-04-14 海能达通信股份有限公司 Identity information processing method and apparatus

Families Citing this family (5)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
CN102916939A (en) * 2012-09-17 2013-02-06 惠州Tcl移动通信有限公司 Implementation method and implementation system of PTT (push-to-talk) call based on VOIP (voice over internet protocol) technology
CN103281309A (en) * 2013-05-09 2013-09-04 厦门亿联网络技术股份有限公司 Broadcasting system based on VOIP (voice over internet phone)
CN107302553B (en) * 2016-04-14 2020-11-06 创新先进技术有限公司 User migration method and device
CN106254966B (en) * 2016-09-14 2019-10-25 深圳市万睿智能科技有限公司 A kind of multiple terminals cell intercom system based on heterogeneous networks medium
CN113099397B (en) * 2021-05-08 2022-05-17 儒安物联科技集团有限公司 Multi-packet voice transmission method based on PTT control, terminal and storage medium

Citations (4)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
CN1780422A (en) * 2004-11-23 2006-05-31 华为技术有限公司 Preemption for group calling service in group telecommunication
CN1801967A (en) * 2005-04-30 2006-07-12 华为技术有限公司 Communication resource distributing method for PTT service in cellular system
CN1917672A (en) * 2005-08-19 2007-02-21 大唐移动通信设备有限公司 Method for controlling speaking right in digital group system, and communication method
CN102916939A (en) * 2012-09-17 2013-02-06 惠州Tcl移动通信有限公司 Implementation method and implementation system of PTT (push-to-talk) call based on VOIP (voice over internet protocol) technology

Family Cites Families (2)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
CN100471290C (en) * 2003-07-19 2009-03-18 华为技术有限公司 Method for implementing half-duplex IP voice communication
AU2003284867A1 (en) * 2003-08-18 2005-03-07 Alcatel Method of voip communication with additional data transmission

Patent Citations (4)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
CN1780422A (en) * 2004-11-23 2006-05-31 华为技术有限公司 Preemption for group calling service in group telecommunication
CN1801967A (en) * 2005-04-30 2006-07-12 华为技术有限公司 Communication resource distributing method for PTT service in cellular system
CN1917672A (en) * 2005-08-19 2007-02-21 大唐移动通信设备有限公司 Method for controlling speaking right in digital group system, and communication method
CN102916939A (en) * 2012-09-17 2013-02-06 惠州Tcl移动通信有限公司 Implementation method and implementation system of PTT (push-to-talk) call based on VOIP (voice over internet protocol) technology

Cited By (1)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
WO2022073219A1 (en) * 2020-10-10 2022-04-14 海能达通信股份有限公司 Identity information processing method and apparatus

Also Published As

Publication number Publication date
CN102916939A (en) 2013-02-06

Similar Documents

Publication Publication Date Title
US7221660B1 (en) System and method for multicast communications using real time transport protocol (RTP)
US10609680B2 (en) Multicast-based group communications in ad hoc arrangements of wireless devices
WO2014040433A1 (en) Implementation method and implementation system for ptt call based on voip technology
US7620413B2 (en) Method for implementing push-to-talk over SIP and multicast RTP related system
US8717949B2 (en) Active speaker identification
US7970425B2 (en) Push-to-talk group call system using CDMA 1x-EVDO cellular network
EP2495911B1 (en) Method and device for uninterruptable wireless group communication sessions
JP2015039194A (en) System and method for adaptive media bundling for voice over internet protocol applications
WO2012079286A1 (en) Method, apparatus and system for remote access to broadcasting
WO2008025230A1 (en) Method and system to achieve cluster communication service
JP2011081816A (en) Method and apparatus for interworking between push-to-talk over cellular (poc) system and instant messaging (im) system
JP2008537648A (en) Method and apparatus for implicit floor control in push talk on cellular systems
WO2014067486A1 (en) Packet forwarding method and relevant device
WO2009071005A1 (en) Method, system, server and client for transmitting media stream data
WO2003013096A1 (en) A speech transmitting method for saving the bandwidth
TWI378684B (en) Communication method and system of internet
WO2013071772A1 (en) Method and device for processing media data packets and conference system
WO2011153780A1 (en) Method, decive and system for controlling connection of group call or of broadcast call
WO2020062530A1 (en) Centerless application software-based communication method and apparatus
CN104580983A (en) Method for achieving video communication PTT
WO2013170812A1 (en) Transmission method for media data stream and thin client
WO2005120103A1 (en) A system of group communication and a method of group call processing based on cdma 2000 high-speed packet data network
WO2014166366A1 (en) Method and device for performing capability negotiation in a long term evolution cluster network
WO2007012265A1 (en) Method and system for realizing apeaking in real time
WO2006116944A1 (en) A method and system for transmitting the media data of the multiparty communication service

Legal Events

Date Code Title Description
121 Ep: the epo has been informed by wipo that ep was designated in this application

Ref document number: 13836867

Country of ref document: EP

Kind code of ref document: A1

NENP Non-entry into the national phase

Ref country code: DE

122 Ep: pct application non-entry in european phase

Ref document number: 13836867

Country of ref document: EP

Kind code of ref document: A1