WO2011000291A1 - Method, device and system for associating real-time transport protocol (rtp) packets in session initiation protocol (sip) session - Google Patents

Method, device and system for associating real-time transport protocol (rtp) packets in session initiation protocol (sip) session Download PDF

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Publication number
WO2011000291A1
WO2011000291A1 PCT/CN2010/074564 CN2010074564W WO2011000291A1 WO 2011000291 A1 WO2011000291 A1 WO 2011000291A1 CN 2010074564 W CN2010074564 W CN 2010074564W WO 2011000291 A1 WO2011000291 A1 WO 2011000291A1
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Prior art keywords
sip
rtp packet
rtp
identifier
session
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PCT/CN2010/074564
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French (fr)
Chinese (zh)
Inventor
张恋
黄蓉军
王明武
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华为技术有限公司
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Publication of WO2011000291A1 publication Critical patent/WO2011000291A1/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
    • H04LTRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
    • H04L65/00Network arrangements, protocols or services for supporting real-time applications in data packet communication
    • H04L65/60Network streaming of media packets
    • H04L65/65Network streaming protocols, e.g. real-time transport protocol [RTP] or real-time control protocol [RTCP]

Definitions

  • the present invention relates to the field of communications, and in particular, to a method, device, and system for associating all RTP packets in a SIP session. Background technique
  • VOIP Voice over Internet Protocol
  • SIP Session Initiation Protocol
  • RTP Real-time Transport Protocol
  • the VOIP recording function that is, the recording server extracts and saves the packets in the voice stream. Specifically, after receiving one SIP signaling, the SIP is parsed first, and the content of the SDP, that is, the addresses of the RTPs, is extracted and saved; then, after the session is established, the data stream of the RTP packet is received and extracted, according to the save. The content matches the RTP, and the data stream of the RTP packet is stored in the corresponding recording file.
  • Embodiments of the present invention provide a method, apparatus, and system for associating an RTP packet in a SIP, which can be associated with a simplified process and improve system performance and stability.
  • a method for associating RTP packets in a SIP session including:
  • An application server including:
  • a receiving unit configured to receive an RTP packet that includes a SIP identifier
  • a parsing unit configured to parse the SIP identifier from the RTP packet
  • An association unit configured to determine, according to the SIP identifier, a SIP connection to which the RTP packet belongs.
  • a method for associating RTP packets in a SIP session including:
  • a terminal comprising:
  • a filling unit configured to fill the SIP identifier into the RTP package
  • a sending unit configured to send the RTP packet.
  • a communication system comprising:
  • a terminal configured to fill a SIP identifier into an RTP packet, and send the RTP packet;
  • an application server configured to receive an RTP packet that includes the SIP identifier, parse the SIP identifier from the RTP packet, and then Determining, according to the SIP identifier, a SIP connection to which the RTP packet belongs.
  • the method, device and system for associating the RTP packet in the SIP tongue are provided by the embodiment of the present invention.
  • the terminal sends the RTP packet
  • the terminal fills the SIP identifier into the RTP packet, and after receiving the RTP packet, the server can pass the packet.
  • the SIP identifier is quickly associated with the SIP to which the RTP packet belongs, avoiding the prior art to parse the SIP and find the IP address and port; then parsing the RTP, finding the IP address and port; and then performing the matching association.
  • P is competing for the complexity of implementation. Also avoid The matching association failure caused by the unfixed IP address and port improves the system performance and stability.
  • FIG. 1 is a flow block diagram of a method for associating an RTP packet in a SIP session according to Embodiment 1 of the present invention
  • FIG. 2 is a structural block diagram of an application server according to Embodiment 2 of the present invention.
  • FIG. 3 is a flow block diagram of a method for associating an RTP packet in a SIP session according to Embodiment 3 of the present invention
  • FIG. 4 is a structural block diagram of a terminal according to Embodiment 4 of the present invention.
  • FIG. 5 is a schematic flowchart of a method for associating an RTP packet in a SIP session according to Embodiment 5 of the present invention
  • FIG. 6 is a schematic structural diagram of a communication system according to Embodiment 6 of the present invention. detailed description
  • a method for associating an RTP packet in a SIP session includes the following steps:
  • S102 Determine, according to the SIP identifier in the RTP packet, the SIP connection to which the RTP packet belongs.
  • the SIP identifier may be a call identifier (Ca ll lD ) of the SIP field or a session identifier (DialogID ) of the SIP field.
  • the method for associating the RTP packet in the SIP session provided by the embodiment of the present invention can quickly match the SIP to which the RTP packet belongs by using the SIP identifier in the packet after the RTP packet is received, thereby avoiding parsing the SIP in the prior art. Find the IP address and port; then parse the RTP, find the IP address and port; then perform the matching association, P strives to reduce the complexity of the implementation. At the same time, the matching association failure caused by the unfixed IP address and port is avoided, which improves the system performance and stability.
  • an application server includes: a receiving unit 201, a parsing unit 202, and an associating unit 203.
  • the receiving unit 201 is configured to receive an RTP packet that includes an identifier of the SIP;
  • the parsing unit 202 is configured to parse the SIP identifier from the RTP packet.
  • the association unit 203 is configured to determine, according to the SIP identifier, a SIP connection to which the RTP packet belongs.
  • the above server further includes:
  • the recording unit 204 is configured to extract the data stream of the RTP packet and store it in the recording file corresponding to the SIP connection to which it belongs. In this way, the association between the RTP packet and the SIP can be utilized to implement the recording function.
  • the application server provided by the embodiment of the present invention can pass the RTP packet after receiving the RTP packet.
  • the SIP identity is quickly associated with the SIP to which the RTP packet belongs. This avoids the prior art to resolve the SIP and find the IP address and port.
  • the RTP is parsed to find the IP address and port. Then the matching association is performed to reduce the association. The complexity of the implementation. At the same time, the matching association failure caused by the unfixed IP address and port is avoided, and the system performance and stability are improved.
  • FIG. 3 A method for associating an RTP packet in a SIP session according to an embodiment of the present invention is as shown in FIG. 3, where the method includes:
  • the filling of the SIP identifier into the RTP packet may include: filling the SIP identifier into the extension header of the RTP; or filling the SIP identifier corresponding mapping table into the RTP. In the extension header.
  • the method for associating the RTP packet in the SIP session provided by the embodiment of the present invention, when the RTP packet is sent, the SIP identifier is filled into the RTP packet.
  • the server can quickly match the SIP to which the RTP packet belongs by using the SIP identifier in the packet, thereby avoiding the prior art to first parse the SIP, find the IP address and port, and then parse the RTP. , find the IP address and port; then perform the matching association, P strives to reduce the complexity of the implementation.
  • the matching association failure caused by the unfixed IP address and port is avoided, and the system performance and stability are improved.
  • the embodiment of the present invention provides a terminal.
  • the terminal includes: a filling unit 401 and a sending unit 402.
  • a filling unit 401 configured to fill the SIP identifier into the RTP packet
  • the sending unit 402 is configured to send the RTP packet.
  • the terminal provided by the embodiment of the present invention fills the SIP identifier into the RTP packet when sending the RTP packet.
  • the server can quickly match the SIP to which the RTP packet belongs by using the SIP identifier in the packet, thereby avoiding the prior art to first parse the SIP, find the IP address and port, and then parse the RTP. , find the IP address and port; then perform matching associations, reducing the complexity of the implementation.
  • the matching association failure caused by the unfixed IP address and port is avoided, and the system performance and stability are improved.
  • the method for associating RTP packets in a SIP session provided by the embodiment of the present invention is described by taking VOIP recording as an example.
  • the RTP stream In a SIP session, after the remote and local calls are established, the RTP stream also establishes two channels. One is local to remote, ⁇ is type A package, and the other is remote to local, assuming type B package. When there are multiple channels, the recording server will capture A1-B, A2-B2, A3-B3, etc. Multi-channel RTP packets.
  • the terminal After the SIP establishes a call, the terminal will identify the SIP identifier, that is, the call identifier of the SIP field. ( Cal l lD ) is populated into the Type A package. Here, the padding field is not limited to the Call l lD. If the session identifier (DialogID) in the service can also identify the session, the DialogID can also be used, and of course, other identifiers are also used. .
  • the RTP protocol defines an extension header, which is generally an extension of the codec. However, this extension information is rarely used in VOIP, so the RTP extension header can be used to populate the SIP identifier Cal l lD.
  • the extension header has 32 bits in total, but it can be divided into two parts, but it can also be used without distinction.
  • the Cal l lD value is too long, and the extended header length is only 32 bits, so you can make a mapping table with Ca 111. D - the corresponding mapping table is filled.
  • the RTP package After the RTP package is packaged, it is sent to the core switch, and the core switch utilizes port mirroring.
  • Port Mirror ing ( Port Mirror ing ) method, mirroring this data to the application server (recording server).
  • the so-called port mirroring method here is a method of mirroring data of one or more ports of a switch to another port or ports.
  • the application server After receiving the RTP packet, the application server can quickly locate the SIP packets in the received A-type packets by parsing the SIP identifier in the RTP packet.
  • the application server extracts the data stream of the RTP packet and stores it in the recording file corresponding to the SIP.
  • A2-B2 and other paths are processed in this way, and the V0IP recording function is realized. The process is shown in Figure 5.
  • the method for associating the RTP packet in the SIP session provided by the embodiment of the present invention, when the terminal sends the RTP packet, the terminal fills the SIP identifier into the RTP packet, and after receiving the RTP packet, the server can quickly pass the SIP identifier in the packet.
  • the SIP to which the RTP packet belongs is matched and matched, so that the SIP is first parsed in the prior art to find the IP address and port; then the RTP is parsed to find the IP address and port;
  • the practice of matching associations, P strives to reduce the complexity of implementation.
  • the matching association failure caused by the unfixed IP address and port is avoided, and the system performance and stability are improved.
  • the matching policy is improved, and the dependence of the recording service on the call control service is reduced in the entire system, and the existing networking and the interaction between the network elements are not changed, so the solution easy and convenient.
  • the communication system provided by the embodiment of the present invention, as shown in FIG. 6, includes: a terminal 601, an application server 602.
  • the terminal 601 is configured to fill the SIP identifier into the RTP packet, and send the RTP packet.
  • the application server 602 is configured to receive the RTP packet that includes the SIP identifier, and parse the SIP identifier from the RTP packet, and then The SIP identity determines the SIP connection to which the RTP packet belongs.
  • the terminal when the terminal sends the RTP packet, the terminal fills the SIP identifier into the RTP packet, and after receiving the RTP packet, the application server can quickly associate with the RTP packet by using the SIP identifier in the packet.
  • SIP performs matching association, avoiding the prior art to parse SIP first, find out the IP address and port; then parse the RTP, find the IP address and port; then perform the matching association, and P strives to reduce the complexity of the implementation.
  • the matching association failure caused by the unfixed IP address and port is avoided, and the system performance and stability are improved.

Abstract

The present invention provides a method, device and system for associating real-time transport protocol (RTP) packets in a session initiation protocol (SIP) session. The present invention is related to the field of communication and is used for associating the RTP packets in the SIP session. The method includes: the RTP packets including the SIP identifiers are received; and the SIP connection which the RTP packets belong to is determined according to the SIP identifiers in the RTP packets. The present invention can simplify the associating process, and improve the system performance and the system stability.

Description

关联 SIP会话中 RTP包的方法、 装置及系统 本申请要求于 2009 年 06 月 29 日提交中国专利局、 申请号为 200910087573. K 发明名称为"关联 SIP ^舌中 RTP包的方法、 装置及系统" 的中国专利申请的优先权, 其全部内容通过引用结合在本申请中。 技术领域  Method, device and system for associating RTP packets in SIP session This application claims to be submitted to the Chinese Patent Office on June 29, 2009, and the application number is 200910087573. K The invention is entitled "Method, device and system for associating SIP^ tongue RTP packets" The priority of the Chinese Patent Application, the entire contents of which is incorporated herein by reference. Technical field
本发明涉及通信领域,尤其涉及一种关联 SIP会话中所有 RTP包的方法、 装置及系统。 背景技术  The present invention relates to the field of communications, and in particular, to a method, device, and system for associating all RTP packets in a SIP session. Background technique
随着互联网的普及, 基于 IP网络的应用越来越广泛, VOIP ( Voice over Internet Protocol,基于 IP网络的语音流 )电话就是其中迅速发展的一项业务。 它将声音采样的信号进行量化, 然后将量化的数据封装成 IP包, 在因特网上 实时传输, 提供比传统业务更多、 更好的服务。  With the popularity of the Internet, applications based on IP networks are becoming more and more widespread. VOIP (Voice over Internet Protocol) is a rapidly growing business. It quantifies the signal sampled by the sound, then encapsulates the quantized data into IP packets and transmits it in real time over the Internet, providing more and better services than traditional services.
在一路会话从开始建立到销毁的过程中, 需要信令进行控制, 对此 SIP ( Session Initiation Protocol, 会话初始协议 )协议提供了很好的支持。 它是 应用层的信令控制协议, 用于创建、 修改和释放一个或多个参与者的会话, 虽然它本身不描述 ^舌消息的内容及其特点, 但它可以通过携带消息体 SDP ( Session Description Protocol,会话描述协议)来描述会话的媒体属性。 SDP 详细描述了会话双方交互协商的过程, 当通话建立 (协商完毕)后, 本地和远 端将会根据 SDP结果,建立 RTP ( Real-time Transport Protocol,实时传输协议 ) 流, 进行通讯交流。  In the process of starting and destroying a session, signaling is required to control, and the SIP (Session Initiation Protocol) protocol is well supported. It is the application layer's signaling control protocol for creating, modifying, and releasing sessions for one or more participants. Although it does not describe the content and characteristics of the message itself, it can carry the message body SDP (Session Description Protocol, a description of the media properties of a session. The SDP describes in detail the process of mutual negotiation between the two parties. After the call is established (negotiated), the local and remote end will establish an RTP (Real-time Transport Protocol) flow according to the SDP result.
VOIP录音功能, 即录音服务器将语音流中的包提取出来并保存。 具体 的, 当接收到一路 SIP信令后, 先将该 SIP解析, 提取出 SDP的内容, 即 RTP 双方的地址, 并保存; 然后在会话建立之后, 接收并提取 RTP包的数据流, 根据保存的内容匹配 RTP, 并将该 RTP包的数据流存入相应的录音文件中。  The VOIP recording function, that is, the recording server extracts and saves the packets in the voice stream. Specifically, after receiving one SIP signaling, the SIP is parsed first, and the content of the SDP, that is, the addresses of the RTPs, is extracted and saved; then, after the session is established, the data stream of the RTP packet is received and extracted, according to the save. The content matches the RTP, and the data stream of the RTP packet is stored in the corresponding recording file.
在实现上述 VOIP录音的过程中, 发明人发现现有技术中至少存在如下 问题:  In the process of realizing the above VOIP recording, the inventors found that at least the following problems exist in the prior art:
现有技术中 RTP和 SIP的匹配关联方法, 非常依赖对 SIP信令及其 SDP的 解析, 但会话初始协议和实时传输协议不是强相关联的协议, 它们之间没有 耦合关系, RTP控制自己的语音流, 它仅受 SIP信令的驱使, 被动地工作。 另夕卜, 在一个完全流程的 SIP会话中, RTP双方的 IP地址和端口, 并不是 一直固定的, 有可能出现 RTP无法匹配关联 SIP的情况。 发明内容 In the prior art, the matching association method between RTP and SIP relies heavily on the resolution of SIP signaling and its SDP, but the session initial protocol and the real-time transmission protocol are not strongly associated protocols, and there is no between them. The coupling relationship, RTP controls its own voice stream, which is driven only by SIP signaling and works passively. In addition, in a full-flow SIP session, the IP addresses and ports of the RTP parties are not always fixed, and there may be cases where the RTP cannot match the associated SIP. Summary of the invention
本发明的实施例提供一种关联 SIP ^舌中 RTP包的方法、 装置及系统, 能够关联简化流程、 提高系统性能和稳定性。  Embodiments of the present invention provide a method, apparatus, and system for associating an RTP packet in a SIP, which can be associated with a simplified process and improve system performance and stability.
为达到上述目的, 本发明的实施例采用如下技术方案:  In order to achieve the above object, the embodiment of the present invention adopts the following technical solutions:
一种关联 SIP会话中 RTP包的方法, 包括:  A method for associating RTP packets in a SIP session, including:
接收含有会话初始协议 SIP标识的实时传输协议 RTP包;  Receiving a real-time transport protocol RTP packet containing a session initial protocol SIP identifier;
据所述 RTP包中的 SIP标识确定所述 RTP包所属的 SIP连接。  Determining, according to the SIP identifier in the RTP packet, the SIP connection to which the RTP packet belongs.
一种应用服务器, 包括:  An application server, including:
接收单元, 用于接收含有 SIP标识的 RTP包;  a receiving unit, configured to receive an RTP packet that includes a SIP identifier;
解析单元, 用于从所述 RTP包中解析出所述 SIP标识;  a parsing unit, configured to parse the SIP identifier from the RTP packet;
关联单元, 用于才 据所述 SIP标识确定所述 RTP包所属的 SIP连接。 一种关联 SIP会话中 RTP包的方法, 包括:  And an association unit, configured to determine, according to the SIP identifier, a SIP connection to which the RTP packet belongs. A method for associating RTP packets in a SIP session, including:
将 SIP标识填充进 RTP包中;  Fill the SIP ID into the RTP package;
发送所述 RTP包。  Send the RTP packet.
一种终端, 包括:  A terminal, comprising:
填充单元, 用于将 SIP标识填充进 RTP包中;  a filling unit, configured to fill the SIP identifier into the RTP package;
发送单元, 用于发送所述 RTP包。  a sending unit, configured to send the RTP packet.
一种通信系统, 包括:  A communication system comprising:
终端, 用于将 SIP标识填充进 RTP包中, 并发送所述 RTP包; 应用服务器, 用于接收含有所述 SIP标识的 RTP包, 从所述 RTP包中 解析出所述 SIP标识,并才 据所述 SIP标识确定所述 RTP包所属的 SIP连接。  a terminal, configured to fill a SIP identifier into an RTP packet, and send the RTP packet; an application server, configured to receive an RTP packet that includes the SIP identifier, parse the SIP identifier from the RTP packet, and then Determining, according to the SIP identifier, a SIP connection to which the RTP packet belongs.
本发明实施例提供的关联 SIP ^舌中 RTP包的方法、装置及系统, 终端 在发送 RTP包时,终端就将 SIP标识填充进 RTP包中,服务器在接收到 RTP 包后, 能够通过包中的 SIP标识迅速与该 RTP包所属的 SIP进行匹配关联, 避免了现有技术中先解析 SIP, 找出 IP地址和端口; 再解析 RTP, 找出 IP 地址和端口; 然后进行匹配关联的做法, P争低了实现的复杂度。 同时也避免 了因 IP地址和端口的不固定所造成的匹配关联失败, 提高了系统性能和稳 定性。 附图说明 The method, device and system for associating the RTP packet in the SIP tongue are provided by the embodiment of the present invention. When the terminal sends the RTP packet, the terminal fills the SIP identifier into the RTP packet, and after receiving the RTP packet, the server can pass the packet. The SIP identifier is quickly associated with the SIP to which the RTP packet belongs, avoiding the prior art to parse the SIP and find the IP address and port; then parsing the RTP, finding the IP address and port; and then performing the matching association. P is competing for the complexity of implementation. Also avoid The matching association failure caused by the unfixed IP address and port improves the system performance and stability. 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一些实施例, 对于本领域普通技术人员来讲, 在不付出创造性劳动性的前提下, 还可以根据这些附图获得其他的附图。  In order to more clearly illustrate the embodiments of the present invention or the technical solutions in the prior art, the drawings used in the embodiments or the description of the prior art will be briefly described below. Obviously, the drawings in the following description are only It is a certain embodiment of the present invention, and other drawings can be obtained from those skilled in the art without any inventive labor.
图 1为本发明实施例 1提供的关联 SIP会话中 RTP包的方法的流程框 图;  1 is a flow block diagram of a method for associating an RTP packet in a SIP session according to Embodiment 1 of the present invention;
图 2为本发明实施例 2提供的应用服务器的结构框图;  2 is a structural block diagram of an application server according to Embodiment 2 of the present invention;
图 3为本发明实施例 3提供的关联 SIP会话中 RTP包的方法的流程框 图;  3 is a flow block diagram of a method for associating an RTP packet in a SIP session according to Embodiment 3 of the present invention;
图 4为本发明实施例 4提供的终端的结构框图;  4 is a structural block diagram of a terminal according to Embodiment 4 of the present invention;
图 5为本发明实施例 5提供的关联 SIP会话中 RTP包的方法的流程示意 图;  FIG. 5 is a schematic flowchart of a method for associating an RTP packet in a SIP session according to Embodiment 5 of the present invention;
图 6为本发明实施例 6提供的通信系统的结构示意图。 具体实施方式  FIG. 6 is a schematic structural diagram of a communication system according to Embodiment 6 of the present invention. detailed description
下面将结合本发明实施例中的附图,对本发明实施例中的技术方案进行 清楚、 完整地描述, 显然, 所描述的实施例仅仅是本发明一部分实施例, 而 不是全部的实施例。 基于本发明中的实施例, 本领域普通技术人员在没有作 出创造性劳动前提下所获得的所有其他实施例, 都属于本发明保护的范围。  The technical solutions in the embodiments of the present invention are clearly and completely described in the following with reference to the accompanying drawings in the embodiments of the present invention. It is obvious that the described embodiments are only a part of the embodiments of the present invention, and not all of the embodiments. All other embodiments obtained by a person of ordinary skill in the art based on the embodiments of the present invention without creative efforts are within the scope of the present invention.
实施例 1:  Example 1:
本发明实施例提供的关联 SIP会话中 RTP包的方法, 如图 1所示, 该方 法步骤包括:  A method for associating an RTP packet in a SIP session, as shown in FIG. 1 , includes the following steps:
S 101、 接收含有会话初始协议 S I P标识的实时传输协议 RTP包; S101. Receive a real-time transport protocol RTP packet that includes a session initial protocol S I P identifier.
S102、 才 据 RTP包中的 SIP标识确定该 RTP包所属的 SIP连接。 S102. Determine, according to the SIP identifier in the RTP packet, the SIP connection to which the RTP packet belongs.
其中, 上述 SIP标识可以为 SIP字段的呼叫标识(Ca l l lD )或 SIP字段 的会话标识(DialogID )。 本发明实施例提供的关联 SIP会话中 RTP包的方法,在接收到 RTP包后, 能够通过包中的 SIP标识迅速与该 RTP包所属的 SIP进行匹配关联,避免了 现有技术中先解析 SIP, 找出 IP地址和端口;再解析 RTP, 找出 IP地址和端 口; 然后进行匹配关联的做法, P争低了实现的复杂度。 同时也避免了因 IP 地址和端口的不固定所造成的匹配关联失败, 提高了系统性能和稳定性。 The SIP identifier may be a call identifier (Ca ll lD ) of the SIP field or a session identifier (DialogID ) of the SIP field. The method for associating the RTP packet in the SIP session provided by the embodiment of the present invention can quickly match the SIP to which the RTP packet belongs by using the SIP identifier in the packet after the RTP packet is received, thereby avoiding parsing the SIP in the prior art. Find the IP address and port; then parse the RTP, find the IP address and port; then perform the matching association, P strives to reduce the complexity of the implementation. At the same time, the matching association failure caused by the unfixed IP address and port is avoided, which improves the system performance and stability.
实施例 2:  Example 2:
针对实施例 1, 本发明实施例提供一种应用服务器, 如图 2所示, 该应 用服务器包括: 接收单元 201、 解析单元 202、 关联单元 203。  For the first embodiment of the present invention, an application server is provided. As shown in FIG. 2, the application server includes: a receiving unit 201, a parsing unit 202, and an associating unit 203.
接收单元 201, 用于接收含有 S I P标识的 RTP包;  The receiving unit 201 is configured to receive an RTP packet that includes an identifier of the SIP;
解析单元 202, 用于从该 RTP包中解析出 SIP标识;  The parsing unit 202 is configured to parse the SIP identifier from the RTP packet.
关联单元 203, 用于根据 SIP标识确定 RTP包所属的 SIP连接。  The association unit 203 is configured to determine, according to the SIP identifier, a SIP connection to which the RTP packet belongs.
进一步地, 上述服务器还包括:  Further, the above server further includes:
录音单元 204, 用于提取 RTP包的数据流, 存入其所属的 SIP连接对应 的录音文件中。 这样, 能够利用 RTP包与 SIP的关联, 实现录音功能。  The recording unit 204 is configured to extract the data stream of the RTP packet and store it in the recording file corresponding to the SIP connection to which it belongs. In this way, the association between the RTP packet and the SIP can be utilized to implement the recording function.
本发明实施例提供的应用服务器, 在接收到 RTP包后, 能够通过包中的 The application server provided by the embodiment of the present invention can pass the RTP packet after receiving the RTP packet.
SIP标识迅速与该 RTP包所属的 SIP进行匹配关联, 避免了现有技术中先解 析 SIP, 找出 IP地址和端口;再解析 RTP, 找出 IP地址和端口; 然后进行匹 配关联的做法, 降低了实现的复杂度。 同时也避免了因 IP地址和端口的不 固定所造成的匹配关联失败, 提高了系统性能和稳定性。 The SIP identity is quickly associated with the SIP to which the RTP packet belongs. This avoids the prior art to resolve the SIP and find the IP address and port. The RTP is parsed to find the IP address and port. Then the matching association is performed to reduce the association. The complexity of the implementation. At the same time, the matching association failure caused by the unfixed IP address and port is avoided, and the system performance and stability are improved.
实施例 3:  Example 3:
本发明实施例提供的关联 SIP会话中 RTP包的方法, 如图 3所示, 该方 法包括:  A method for associating an RTP packet in a SIP session according to an embodiment of the present invention is as shown in FIG. 3, where the method includes:
5301、 将 SIP标识填充进 RTP包中;  5301. Fill the SIP identifier into the RTP package.
5302、 发送该 RTP包。  5302. Send the RTP packet.
进一步地, 步骤 S301中, 将 SIP标识填充进 RTP包中可以具体包括: 将 SIP标识填充进 RTP的扩展头中;或者,将 SIP标识对应映射表填充进 RTP 的扩展头中。 Further, in step S301, the filling of the SIP identifier into the RTP packet may include: filling the SIP identifier into the extension header of the RTP; or filling the SIP identifier corresponding mapping table into the RTP. In the extension header.
本发明实施例提供的关联 SIP会话中 RTP包的方法, 在发送 RTP包时, 就将 SIP标识填充进该 RTP包中。 这样, 使得服务器在接收到 RTP包后, 能 够通过包中的 SIP标识迅速与该 RTP包所属的 SIP进行匹配关联,避免了现 有技术中先解析 SIP,找出 IP地址和端口;再解析 RTP,找出 IP地址和端口; 然后进行匹配关联的做法, P争低了实现的复杂度。 同时也避免了因 IP地址 和端口的不固定所造成的匹配关联失败, 提高了系统性能和稳定性。  The method for associating the RTP packet in the SIP session provided by the embodiment of the present invention, when the RTP packet is sent, the SIP identifier is filled into the RTP packet. In this way, after receiving the RTP packet, the server can quickly match the SIP to which the RTP packet belongs by using the SIP identifier in the packet, thereby avoiding the prior art to first parse the SIP, find the IP address and port, and then parse the RTP. , find the IP address and port; then perform the matching association, P strives to reduce the complexity of the implementation. At the same time, the matching association failure caused by the unfixed IP address and port is avoided, and the system performance and stability are improved.
实施例 4:  Example 4:
针对实施例 3, 本发明实施例提供一种终端, 如图 4所示, 该终端包括: 填充单元 401、 发送单元 402。  For the embodiment 3, the embodiment of the present invention provides a terminal. As shown in FIG. 4, the terminal includes: a filling unit 401 and a sending unit 402.
填充单元 401, 用于将 SIP标识填充进 RTP包中;  a filling unit 401, configured to fill the SIP identifier into the RTP packet;
发送单元 402, 用于发送该 RTP包。  The sending unit 402 is configured to send the RTP packet.
本发明实施例提供的终端,在发送 RTP包时,就将 SIP标识填充进该 RTP 包中。 这样, 使得服务器在接收到 RTP包后, 能够通过包中的 SIP标识迅速 与该 RTP包所属的 SIP进行匹配关联, 避免了现有技术中先解析 SIP, 找出 IP地址和端口;再解析 RTP,找出 IP地址和端口;然后进行匹配关联的做法, 降低了实现的复杂度。 同时也避免了因 IP地址和端口的不固定所造成的匹 配关联失败, 提高了系统性能和稳定性。  The terminal provided by the embodiment of the present invention fills the SIP identifier into the RTP packet when sending the RTP packet. In this way, after receiving the RTP packet, the server can quickly match the SIP to which the RTP packet belongs by using the SIP identifier in the packet, thereby avoiding the prior art to first parse the SIP, find the IP address and port, and then parse the RTP. , find the IP address and port; then perform matching associations, reducing the complexity of the implementation. At the same time, the matching association failure caused by the unfixed IP address and port is avoided, and the system performance and stability are improved.
实施例 5:  Example 5
本发明实施例提供的关联 SIP会话中 RTP包的方法, 以实现 VOIP录音 为例进行说明。  The method for associating RTP packets in a SIP session provided by the embodiment of the present invention is described by taking VOIP recording as an example.
在一个 SIP会话中,远端和本地建立通话后, RTP流也建立了两条通道。 一条是本地到远端的, 殳是 A类型包, 另一条是远端到本地的, 假设是 B 类型包。 当有多路时, 录音服务器上将会抓取到 A1- Bl, A2- B2, A3- B3,等等 多路的 RTP数据包。  In a SIP session, after the remote and local calls are established, the RTP stream also establishes two channels. One is local to remote, 殳 is type A package, and the other is remote to local, assuming type B package. When there are multiple channels, the recording server will capture A1-B, A2-B2, A3-B3, etc. Multi-channel RTP packets.
在 SIP建立通话后, 终端将一路 SIP标识, 即将 SIP字段的呼叫标识 ( Cal l lD )填充到 A类型包中。 此处, 填充字段并不局限于 Cal l lD, 如果业 务中会话标识(DialogID )也能标识该路会话, 使用 DialogID也可以, 当 然, 也可以是其他标识, 本发明实施例在此不做限定。 After the SIP establishes a call, the terminal will identify the SIP identifier, that is, the call identifier of the SIP field. ( Cal l lD ) is populated into the Type A package. Here, the padding field is not limited to the Call l lD. If the session identifier (DialogID) in the service can also identify the session, the DialogID can also be used, and of course, other identifiers are also used. .
在终端 A侧, 发送 RTP包时, 需要对 RTP包进行打包。 RTP协议定义了 一个扩展头, 它一般是对编解码的扩充信息, 但这个扩展信息在 VOIP中几 乎没有使用, 所以可以利用 RTP扩展头, 将 SIP标识 Cal l lD填充进去。  On the terminal A side, when sending RTP packets, the RTP packets need to be packaged. The RTP protocol defines an extension header, which is generally an extension of the codec. However, this extension information is rarely used in VOIP, so the RTP extension header can be used to populate the SIP identifier Cal l lD.
扩展头共 32位, 分两部分, 但也可以不区分这两部分, 全部加以利用, 考虑到 Cal l lD值太长, 而扩展头长度只有 32位, 所以可以做一个映射表, 用 Ca 111 D——对应的映射表进行填充。  The extension header has 32 bits in total, but it can be divided into two parts, but it can also be used without distinction. The Cal l lD value is too long, and the extended header length is only 32 bits, so you can make a mapping table with Ca 111. D - the corresponding mapping table is filled.
对 RTP包进行打包好后, 发送给核心交换机, 核心交换机利用端口镜像 After the RTP package is packaged, it is sent to the core switch, and the core switch utilizes port mirroring.
( Port Mirror ing ) 方法, 将这些数据镜像到应用服务器(录音服务器)。 这里所谓的端口镜像方法, 就是把交换机一个或多个端口的数据镜像到另外 的一个或多个端口的方法。 ( Port Mirror ing ) method, mirroring this data to the application server (recording server). The so-called port mirroring method here is a method of mirroring data of one or more ports of a switch to another port or ports.
应用服务器在接收到 RTP包后, 通过解析 RTP包中的 SIP标识, 可以非 常方便的快速定位收到的这些 A类型包是哪路 SIP会话中的。  After receiving the RTP packet, the application server can quickly locate the SIP packets in the received A-type packets by parsing the SIP identifier in the RTP packet.
如前所述, RTP流正常通话后, 建立了两条通道, 是一来一往的, 很容 易由 A类型的包推导出 B类型的包。 因为在双工情况下, A类型和 B类型的 包, 他们的 RTP流地址是反向对称的, 即 A包的源地址、 目标地址, 和 B包 的源地址、 目标地址, 只是信息反过来而已, 内容是相同的。  As mentioned above, after the RTP stream is in normal conversation, two channels are established, which are one-to-one. It is easy to derive the B-type packet from the A-type packet. Because in the case of duplex, type A and type B packets, their RTP stream addresses are reverse symmetric, that is, the source address of the A packet, the destination address, and the source address and destination address of the B packet, but the information is reversed. The content is the same.
然后, 应用服务器提取出该 RTP包的数据流, 存入所属 SIP对应的录音 文件中, 同理, A2- B2以及其他各路都以这种方式处理, 实现了 V0IP的录音 功能。 其过程如图 5所示。  Then, the application server extracts the data stream of the RTP packet and stores it in the recording file corresponding to the SIP. Similarly, A2-B2 and other paths are processed in this way, and the V0IP recording function is realized. The process is shown in Figure 5.
本发明实施例提供的关联 SIP会话中 RTP包的方法, 终端在发送 RTP包 时, 就将 SIP标识填充进 RTP包中, 服务器在接收到 RTP包后, 能够通过包 中的 SIP标识迅速与该 RTP包所属的 SIP进行匹配关联,避免了现有技术中 先解析 SIP, 找出 IP地址和端口; 再解析 RTP, 找出 IP地址和端口; 然后 进行匹配关联的做法, P争低了实现的复杂度。 同时也避免了因 IP地址和端 口的不固定所造成的匹配关联失败, 提高了系统性能和稳定性。 另外, 本发 明实施例由于改进了匹配策略, 在整个系统中, 减少了录音业务对呼叫控制 业务上的依赖, 且对现有的组网, 以及网元之间的交互都没有改变, 所以方 案简单方便。 The method for associating the RTP packet in the SIP session provided by the embodiment of the present invention, when the terminal sends the RTP packet, the terminal fills the SIP identifier into the RTP packet, and after receiving the RTP packet, the server can quickly pass the SIP identifier in the packet. The SIP to which the RTP packet belongs is matched and matched, so that the SIP is first parsed in the prior art to find the IP address and port; then the RTP is parsed to find the IP address and port; The practice of matching associations, P strives to reduce the complexity of implementation. At the same time, the matching association failure caused by the unfixed IP address and port is avoided, and the system performance and stability are improved. In addition, in the embodiment of the present invention, the matching policy is improved, and the dependence of the recording service on the call control service is reduced in the entire system, and the existing networking and the interaction between the network elements are not changed, so the solution easy and convenient.
实施例 6:  Example 6:
本发明实施例提供的通信系统, 如图 6所示, 该系统包括: 终端 601, 应用服务器 602。  The communication system provided by the embodiment of the present invention, as shown in FIG. 6, includes: a terminal 601, an application server 602.
终端 601, 用于将 SIP标识填充进 RTP包中, 并发送该 RTP包; 应用服务器 602, 用于接收含有该 SIP标识的 RTP包, 从 RTP包中解析 出该 SIP标识, 并才艮据该 SIP标识确定该 RTP包所属的 SIP连接。  The terminal 601 is configured to fill the SIP identifier into the RTP packet, and send the RTP packet. The application server 602 is configured to receive the RTP packet that includes the SIP identifier, and parse the SIP identifier from the RTP packet, and then The SIP identity determines the SIP connection to which the RTP packet belongs.
本发明实施例提供的通信系统, 终端在发送 RTP包时, 终端就将 SIP标 识填充进 RTP包中, 应用服务器在接收到 RTP包后, 能够通过包中的 SIP标 识迅速与该 RTP包所属的 SIP进行匹配关联,避免了现有技术中先解析 SIP, 找出 IP地址和端口; 再解析 RTP, 找出 IP地址和端口; 然后进行匹配关联 的做法, P争低了实现的复杂度。 同时也避免了因 IP地址和端口的不固定所 造成的匹配关联失败, 提高了系统性能和稳定性。  In the communication system provided by the embodiment of the present invention, when the terminal sends the RTP packet, the terminal fills the SIP identifier into the RTP packet, and after receiving the RTP packet, the application server can quickly associate with the RTP packet by using the SIP identifier in the packet. SIP performs matching association, avoiding the prior art to parse SIP first, find out the IP address and port; then parse the RTP, find the IP address and port; then perform the matching association, and P strives to reduce the complexity of the implementation. At the same time, the matching association failure caused by the unfixed IP address and port is avoided, and the system performance and stability are improved.
本领域普通技术人员可以理解实现上述实施例方法中的全部或部分步 骤是可以通过程序来指令相关的硬件完成,相关程序可以存储于一种计算机 可读存储介质中, 所述存储介质可以是只读存储器, 磁盘或光盘等。  A person skilled in the art can understand that all or part of the steps of implementing the foregoing embodiments may be performed by a program to instruct related hardware, and the related program may be stored in a computer readable storage medium, and the storage medium may be Read memory, disk or CD, etc.
以上所述, 仅为本发明的具体实施方式, 但本发明的保护范围并不局限 于此, 任何熟悉本技术领域的技术人员在本发明揭露的技术范围内, 可轻易 想到变化或替换, 都应涵盖在本发明的保护范围之内。 因此, 本发明的保护 范围应所述以权利要求的保护范围为准。  The above is only the specific embodiment of the present invention, but the scope of the present invention is not limited thereto, and any person skilled in the art can easily think of changes or substitutions within the technical scope of the present invention. It should be covered by the scope of the present invention. Therefore, the scope of the invention should be determined by the scope of the claims.

Claims

权 利 要 求 Rights request
1、 一种关联 SIP会话中 RTP包的方法, 其特征在于, 包括:  A method for associating an RTP packet in a SIP session, the method comprising:
接收含有会话初始协议 SIP标识的实时传输协议 RTP包;  Receiving a real-time transport protocol RTP packet containing a session initial protocol SIP identifier;
根据所述 RTP包中的 SIP标识确定所述 RTP包所属的 SIP连接。  Determining, according to the SIP identifier in the RTP packet, a SIP connection to which the RTP packet belongs.
2、 根据权利要求 1所述的方法, 其特征在于, 所述根据所述 RTP包中 的 S IP标识确定所述 RTP包所属的 SIP连接之后, 还包括:  The method according to claim 1, wherein after determining the SIP connection to which the RTP packet belongs according to the S IP identifier in the RTP packet, the method further includes:
提取 RTP包的数据流, 存入其所属的 SIP连接对应的录音文件中。  The data stream of the RTP packet is extracted and stored in the recording file corresponding to the SIP connection to which it belongs.
3、 根据权利要求 1所述的方法, 其特征在于, 所述 SIP标识是由终端 在发送 RTP包时, 填充进所述 RTP包中的。  The method according to claim 1, wherein the SIP identifier is filled in the RTP packet by the terminal when transmitting the RTP packet.
4、 根据权利要求 1所述的方法, 其特征在于, 所述方法还包括: 才艮据本地 RTP包的地址, 反向查找出同一会话中远端的 RTP包。  The method according to claim 1, wherein the method further comprises: searching for the remote RTP packet in the same session in reverse according to the address of the local RTP packet.
5、 根据权利要求 1至 4任意一个所述的方法, 其特征在于, 所述 SIP 标识包括:  The method according to any one of claims 1 to 4, wherein the SIP identifier comprises:
SIP字段的呼叫标识或 SIP字段的会话标识。  The call ID of the SIP field or the session ID of the SIP field.
6、 一种应用服务器, 其特征在于, 包括:  6. An application server, comprising:
接收单元, 用于接收含有 SIP标识的 RTP包;  a receiving unit, configured to receive an RTP packet that includes a SIP identifier;
解析单元, 用于从所述 RTP包中解析出所述 SIP标识;  a parsing unit, configured to parse the SIP identifier from the RTP packet;
关联单元, 用于根据所述 SIP标识确定所述 RTP包所属的 SIP连接。 And an association unit, configured to determine, according to the SIP identifier, a SIP connection to which the RTP packet belongs.
7、 根据权利要求 6所述的应用服务器, 其特征在于, 所述应用服务器 还包括: The application server according to claim 6, wherein the application server further comprises:
录音单元, 用于提取 RTP包的数据流, 存入其所属的 SIP连接对应的录 音文件中。  The recording unit is configured to extract the data stream of the RTP packet and store it in the recording file corresponding to the SIP connection to which it belongs.
8、 一种关联 SIP会话中 RTP包的方法, 其特征在于, 包括:  8. A method for associating an RTP packet in a SIP session, the method comprising:
将 S I P标识填充进 RTP包中;  Fill the S I P identifier into the RTP packet;
发送所述 RTP包。  Send the RTP packet.
9、 根据权利要求 8所述的方法, 其特征在于, 所述将 SIP标识填充进 RTP包中, 包括: 9. The method according to claim 8, wherein the filling the SIP identifier into In the RTP package, it includes:
将所述 SIP标识填充进所述 RTP的扩展头中; 或者  Filling the SIP identifier into the extension header of the RTP; or
将所述 SIP标识对应映射表填充进所述 RTP的扩展头中。  Filling the SIP identity mapping table into the extension header of the RTP.
10、 一种终端, 其特征在于, 包括:  10. A terminal, comprising:
填充单元, 用于将 SIP标识填充进 RTP包中;  a filling unit, configured to fill the SIP identifier into the RTP package;
发送单元, 用于发送所述 RTP包。  a sending unit, configured to send the RTP packet.
11、 一种通信系统, 其特征在于, 包括:  A communication system, comprising:
终端, 用于将 SIP标识填充进 RTP包中, 并发送所述 RTP包; 应用服务器, 用于接收含有所述 SIP标识的 RTP包, 从所述 RTP包中解 析出所述 SIP标识, 并根据所述 SIP标识确定所述 RTP包所属的 SIP连接。  a terminal, configured to fill a SIP identifier into an RTP packet, and send the RTP packet; an application server, configured to receive an RTP packet that includes the SIP identifier, parse the SIP identifier from the RTP packet, and The SIP identifier determines a SIP connection to which the RTP packet belongs.
PCT/CN2010/074564 2009-06-29 2010-06-28 Method, device and system for associating real-time transport protocol (rtp) packets in session initiation protocol (sip) session WO2011000291A1 (en)

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