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Publication numberCN101127712 B
Publication typeGrant
Application numberCN 200710143006
Publication date25 May 2011
Filing date20 Aug 2007
Priority date20 Aug 2007
Also published asCN101127712A
Publication number200710143006.4, CN 101127712 B, CN 101127712B, CN 200710143006, CN-B-101127712, CN101127712 B, CN101127712B, CN200710143006, CN200710143006.4
Inventors周蕙菁, 张新林
Applicant中兴通讯股份有限公司
Export CitationBiBTeX, EndNote, RefMan
External Links: SIPO, Espacenet
A method for solving synchronization source identity confliction in RTP session
CN 101127712 B
Abstract
The invention discloses a method to solve the conflict of the synchronous source labels in the RTP conversation. After building the RTP conversation, if a participator I in the conversation finds the self-synchronous source label has conflict with the synchronous source label of a new participator II, the synchronous source label of the participator I is updated immediately and the message is transmitted continuously; the synchronous source label of the participator II is updated immediately when the participator II finds the self- synchronous source label has conflict with the synchronous source label of the participator I and continues the message transmission; when finding that the synchronous source label of the participator II has conflict with the synchronous source label of the participator I, the other participators in the conversation directly discard the message from the participator II and continue to receive the message from the participator I. The invention can solve the SSRC conflict in the RTP conversation without starting the RTCP, ensure that the current connected voice is not impacted during the SSRC conflict through analyzing the Sequence Number in the message and recover the conversation in a short period.
Claims(7)  translated from Chinese
1. 一种解决RTP会话中同步源标识冲突的方法,其特征在于,在建立RTP会话后,如果该会话中的一个参与者一发现自己的同步源标识与新加入该会话的参与者二的同步源标识发生冲突,则立即更新自己的同步源标识,并继续发送报文;所述参与者二在发现自己的同步源标识与所述参与者一发生冲突时也立即更新其同步源标识,并继续发送报文;该会话中的其它参与者在发现参与者二的同步源标识与参与者一发生冲突时,直接丢弃所述参与者二的报文,继续接收参与者一的报文。 A workaround RTP session synchronization source identity conflict, characterized in that, after the establishment of RTP session if the session a participant finds himself a synchronization source identification and new participants to join the session II synchronization source identity conflict, immediately update their synchronization source identification, and continues to send the message; the two participants also immediately update its synchronization source identification found their synchronization source identifier and the participant a violation occurs, and continues to send the message; the other participants in the session when participants found two synchronized source identification of a conflict with the participant, the participant directly discard two packets, continues to receive a packet of participants.
2.如权利要求1所述的方法,其特征在于,所述参与者一在更新自己的同步源标识后,按照已有的频率继续发送报文,所述会话中的其它参与者通过分析所接收的报文中的序列号得到所述参与者一发送的报文。 2. The method according to claim 1, characterized in that said one participant in the updated own synchronization source identification, in accordance with the existing frequency continues to send messages to other participants in the session through the analysis the received packet sequence number of the participants receive a packet sent.
3.如权利要求1所述的方法,其特征在于,在所述参与者一与所述参与者二均立即更新同步源标识后,所述参与者一及所述会话中的其它参与者即可正常接收所述参与者二发送的报文。 3. The method according to claim 1, characterized in that, after one of the two participants are immediately updated with the synchronized source identification participant, one of the participants of the session and the other participants i.e. Participants can normally receive the second transmission of packets.
4.如权利要求1或2或3所述的方法,其特征在于,所述参与者二加入所述会话后,开始发送报文,所述参与者一是在收到所述参与者二发送的所述报文后,发现自己的同步源标识与所述参与者二发生冲突。 4. The method of 1 or 2 or 3 of the preceding claims, wherein the second participant to join the session, start sending messages, one of the participants in the two sending participant received After the message, they found their synchronization source identifier and the participant two conflict.
5.如权利要求4所述的方法,其特征在于,所述参与者二是在收到所述参与者一发送的报文后,发现自己的同步源标识与所述参与者一发生冲突。 5. The method according to claim 4, wherein the second is the participant after receiving a message sent by the participant, found their synchronization source identifier and the participants of a conflict.
6.如权利要求5所述的方法,其特征在于,所述建立RTP会话的参与者最初有两个或更多。 6. The method according to claim 5, characterized in that said first RTP session participant to establish two or more.
7.如权利要求1所述的方法,其特征在于,所述同步源标识的更新是由实时传输协议完成。 7. The method according to claim 1, wherein said synchronization source identification is done by updating the real-time transport protocol.
Description  translated from Chinese

一种解决RTP会话中同步源标识冲突的方法 One way RTP session synchronization source identified conflict resolution

技术领域 Technical Field

[0001]本发明属于 VoIP(Voice over Internet Protocol,IP i吾音技术)技术领域, 具体涉及一种在RTCP (RTP Contro 1 Protoco 1,RTP控制协议)不启动的情况下,解决RTP (Real-Time Transport Protocol,实时传输协议)会话中SSRC(Synchronization source,同步源标识)冲突的方法。 [0001] The present invention pertains to VoIP (Voice over Internet Protocol, IP i I sound technology) technology, specifically relates to a case of RTCP (RTP Contro 1 Protoco 1, RTP Control Protocol) does not start to address the RTP (Real- Time Transport Protocol, Real-time Transport Protocol) session SSRC (Synchronization source, synchronization source identification) method of conflict.

背景技术 Background

[0002] RTP为实时数据提供端到端的传输功能,如交互的音频视频数据。 [0002] RTP provides end-to-real-time data transfer capabilities, such as interactive audio and video data. 功能包括载荷类型辨别,序列号编码,时戳,传输监控。 Features include payload type identification, serial number coding, time stamp, and video surveillance. RTP在UDP^ser DatagramProtocol,用户数据报协议)协议之上,利用UDP复用,包校验功能协作共同完成传输层功能。 RTP over UDP ^ ser DatagramProtocol, User Datagram Protocol), the use of UDP multiplexing, packet checksum function Collaboration complete transport layer functions.

[0003] RTP协议包括两部分:RTP(用于传输实时数据)和RTCP(用于RTP业务质量监控)会话信息交互。 [0003] RTP protocol consists of two parts: RTP (for the transmission of real-time data) and RTCP (RTP for service quality monitoring) session information exchange. RTP部分主要完成载荷封装,序列号管理,时戳编码,同步源标识符功能。 RTP payload package complete part number, serial number management, time stamp coding, synchronization source identifier function. RTCP周期地向RTP会话的各方发送控制报文。 Periodically transmits RTCP control packets to all parties RTP session. RTCP实现的主要功能是:提供数据传输质量的反馈;这也就是RTP作为传输层协议提供流控、拥塞控制的一个部分,这个反馈可用于自适应编码的控制,用于监控本地或远端错误,对于IP组播可提供给第三方作网络监控用。 The main function of RTCP implementations are: to provide feedback on the quality of data transmission; this is the RTP as a transport protocol to provide flow control, congestion control part, this feedback can be used for adaptive control of coding used to monitor local or remote error For IP multicast is available to third parties for network monitoring and control. RTCP报文分为SR(发送者报告)、RR(接收者报告)、SDES (信源说明)、BYE (会话结束)、APP(特定应用报文)五种类型。 RTCP packets into SR (Sender Report), RR (receiver report), SDES (Source Description), BYE (session ends), APP (application-specific messages) five types.

[0004] RTP报文格式如图1所示,其中 [0004] 1 RTP packet format, wherein

[0005] 版本号V(Version) :2位,代表RTP的版本号; [0005] The version number V (Version): 2 position, on behalf of RTP version number;

[0006] 填充标识PO^dding) :1位,当P = 1时表示包含一个或多个填充字节以进行32 位对齐; [0006] filling identified PO ^ dding): 1 position, when P = 1 means comprises one or more padding bytes for 32-bit aligned;

[0007] 扩展位X(EXtension) :1位,表示是否包括扩展头部; [0007] extension bit X (EXtension): 1 bit that indicates whether or not including the extension of the head;

[0008] CSRC 计数器CC (CSRC Count) :4 位,表示CSRC 的计数(CSRC Count); [0008] CSRC counter CC (CSRC Count): 4, indicates that the CSRC count (CSRC Count);

[0009] 标记位M(Marker) :1位,由Profile文件定义,允许重要事件如帧边界在数据包流中进行标记; [0009] The flag M (Marker): 1 bit defined by the Profile file, allowing significant events such as frame boundaries marked in the packet stream;

[0010] 载荷类型PT (Payload Type) :7位,标识了RTP载荷的格式,它决定了应用程序如何对载荷解码; [0010] The payload type PT (Payload Type): 7 bits identifies the RTP payload format, which determines how the application of the load decoding;

[0011] 序列号(kquence Number) : 16位,用来检测RTP报文丢失和报文乱序时排序的参考,发送方如一个SSRC每发出一个RTP报文,SequenceNumber增加Sequence Number的初始值随机产生,接收方可以由该域检测包的丢失并恢复数据包序列; [0011] Serial Number (kquence Number): 16 bits used to detect when the RTP packet loss and packets out of order sorting reference, the sender as an SSRC issue an RTP packet each, SequenceNumber increase the initial value of Random Sequence Number produce, the receiver can detect the loss of the domain package and restore data packet sequence;

[0012] 时间戳(Timestamp) :32位,记录数据包中数据的第一个字节的采样时刻; [0012] timestamp (Timestamp): 32 bits of the data packets recorded sampling instant of the first byte;

[0013] 同步源标识(SSRC) :32位,标识一个RTP数据流的源,SSRC的产生是随机的,这样,也就有可能在有多个参与者的时候,发生冲突; [0013] synchronization source identification (SSRC): 32 bits identifies a source of the RTP data stream, SSRC is randomly generated, so that it is possible to have more than one participant, when conflict;

[0014] 贡献源标识(CSRC,Contributing Source) :0〜15项,每项32位用于识别该RTP 数据包中的有效载荷的贡献源。 [0014] Contribution source identification (CSRC, Contributing Source): 0~15 item 32 for identifying each packet in the RTP payload source contribution.

[0015] RFC3550是RTP协议的标准。 [0015] RFC3550 standard RTP protocol. 此标准有如下一些定义:[0016] 1,在同一个RTP会话中,所有参与者的SSRC字段要唯一;如果一个RTP会话的参与者发现SSRC冲突(即在一个RTP会话中,参与者的SSRC不唯一),通过发送RTCP BYE报文,退出会话,然后重新生成一个SSRC标识,加入会话; This standard has the following definitions: [0016] 1, in the same RTP session, SSRC field for all participants to be unique; if an RTP session participants found SSRC conflict (ie an RTP session, the participants in the SSRC not unique), by sending RTCP BYE packets, exit the session, then rebuild a SSRC identifier to join the session;

[0017] 2,在多方会话时,如果某一个参与者发现RTP会话另外两个参与者的SSRC冲突, 则会尽可能的通过判断网络传输层地址或者判断RTCP SDES中的内容来区分报文,直到SSRC不再冲突; [0017] 2, in the multi-party session, if one participant found RTP SSRC conflict two other participants in the session, the packets will be distinguished by judging the network transport layer address or judgment RTCP SDES the content as much as possible, SSRC until no conflict;

[0018] 3,在一个RTP会话中,一个新的SSRC被其它参与者认为是有效的数据流的条件是:至少收到2个从这个SSRC发出的连续的RTP报文。 [0018] 3, in an RTP session, a new SSRC by other participants considered valid data flow condition is: received at least two from the SSRC issued consecutive RTP packets.

[0019] 虽然RFC3550有这些定义,但是,RTCP并不是必须启用的。 [0019] Although RFC3550 these definitions, however, RTCP not be enabled. 不启用RTCP可以带来的好处是:节约链路带宽,提高语音质量,有时甚至是还没有实现RTCP的功能。 You can not enable RTCP benefits are: saving link bandwidth, improve voice quality, and sometimes not implemented RTCP function. 在这种情况下(RTCP不启用),如果出现了SSRC冲突,如何解决,协议并没有说明。 In this case (RTCP not enabled), if there is a SSRC conflict, how to resolve, the agreement did not specify.

发明内容 DISCLOSURE

[0020] 本发明要解决的技术问题是提供一种在RTCP不启动的情况下,解决RTP会话中SSRC冲突的方法。 [0020] The technical problem to be solved by the present invention to provide a method in the case of RTCP does not start to solve the conflict SSRC RTP session.

[0021] 为了解决上述问题,本发明提供了一种解决RTP会话中同步源标识冲突的方法, 在建立RTP会话后,如果该会话中的一个参与者一发现自己的同步源标识与新加入该会话的参与者二的同步源标识发生冲突,则立即更新自己的同步源标识,并继续发送报文;所述参与者二在发现自己的同步源标识与所述参与者一发生冲突时也立即更新其同步源标识, 并继续发送报文;该会话中的其它参与者在发现参与者二的同步源标识与参与者一发生冲突时,直接丢弃所述参与者二的报文,继续接收参与者一的报文。 [0021] In order to solve the above problems, the present invention provides an RTP session synchronization source identified ways to solve the conflict, in the establishment of RTP session if the session a participant finds himself a synchronization source identification and new entrants to the synchronization source identifying the session participants two clashed immediately update their synchronization source identification, and continues to send the message; the participants find themselves second in synchronization source identification and the participants also immediately when a conflict occurs update its synchronization source identification, and continues to send the message; the other participants in the session when participants found two synchronized source identification of a conflict with the participant, the participant directly discard two packets, continues to receive participation by a message.

[0022] 进一步地,上述的方法还可以具有如下特点:所述参与者一在更新自己的同步源标识后,按照已有的频率继续发送报文,所述会话中的其它参与者通过分析所接收的报文中的序列号得到所述参与者一发送的报文。 [0022] Further, the above method may also have the following characteristics: a participant in the updated own synchronization source identification, in accordance with the existing frequency continues to send messages to other participants in the session through the analysis the received packet sequence number of the participants receive a packet sent.

[0023] 进一步地,上述的方法还可以具有如下特点:在所述参与者一与所述参与者二均立即更新同步源标识后,所述参与者一及所述会话中的其它参与者即可正常接收所述参与者二发送的报文。 [0023] Further, the above method may also have the following characteristics: a participant after the two were immediately update synchronization source identification and the participant, the participant one and the other participants in the session that is Participants can normally receive the second transmission of packets.

[0024] 进一步地,上述的方法还可以具有如下特点:所述参与者二加入所述会话后,开始发送报文,所述参与者一是在收到所述参与者二发送的所述报文后,发现自己的同步源标识与所述参与者二发生冲突。 [0024] Further, the above method may also have the following characteristics: the trailing two participants join the session, start sending messages, one of the participants in receipt of the message sent by the second participant later, they found their synchronization source identifier and the participant two conflict.

[0025] 进一步地,上述的方法还可以具有如下特点:所述参与者二是在收到所述参与者一发送的报文后,发现自己的同步源标识与所述参与者一发生冲突。 [0025] Further, the above method may also have the following characteristics: the participant Second, after receiving a message sent by the participants, found their synchronization source identifier and the participants of a conflict.

[0026] 进一步地,上述的方法还可以具有如下特点:所述建立RTP会话的参与者最初有两个或更多。 [0026] Further, the above method may also have the following characteristics: the establishment of the initial RTP session participant has two or more.

[0027] 进一步地,上述的方法还可以具有如下特点:所述同步源标识的更新是由实时传输协议完成。 [0027] Further, the above method may also have the following characteristics: updating the synchronization source identifier is performed by the real-time transport protocol.

[0028] 通过本发明的方法可以很快消除RTP会话中的SSRC冲突,并且通过对报文中的Sequence Number的分析,保证了在SSRC发生冲突的时候已建立连接的语音不受影响,在发生冲突的SSRC均立即更新之后,只需短暂的时间就可以恢复通话。 [0028] The method of the present invention can quickly eliminate conflict SSRC RTP session, and by telegram Sequence Number of analyzes to ensure that when the SSRC conflict affected voice connection has been established, in the event of immediately after the conflict SSRC are updated only a short time to be able to resume the call. 附图说明 Brief Description

[0029] 图1是RTP报文格式; [0029] FIG. 1 is a RTP packet format;

[0030] 图2是本发明实施例多个参与者进行RTP会话的示意图。 [0030] FIG. 2 is a schematic view of a plurality of participants embodiment of the present invention RTP session. 具体实施方式 DETAILED DESCRIPTION

[0031] 本发明解决SSRC冲突(不启用RTCP)的主要思想是:一旦发现冲突,参与者立即更新自己的SSRC标识,并通过判断发送频率或者序列号字段,尽可能地保证RTP报文的不丢失。 The main idea of [0031] The present invention solves the SSRC conflict (not enabled RTCP) is: once a conflict is found, participants update their SSRC immediately identified, and by judging whether the transmission frequency or a sequence number field, as far as possible to ensure that RTP packets not lost.

[0032] 由于RTP只传送实时数据,本身并不提供任何保证实时传送数据和服务质量的能力,因此,本发明通过RTP提供的序列号信息,在接收端根据报文中的这些信息来判断和接收正确的报文数据。 [0032] Since the RTP only transmit real-time data, does not provide any real-time transmission of data and the ability to ensure quality of service and, therefore, the present invention is provided by the serial number of RTP, on the receiving end of the packet of information to determine and receive the correct message data.

[0033] 下面结合附图及具体实施例对本发明作进一步详细描述。 [0033] below in conjunction with the accompanying drawings and the specific embodiments of the present invention will be further described in detail.

[0034] 参见图2,A,B, C三个参与者通过IP网络连接,进行RTP会话。 [0034] 2, A, B, C are connected by three participants see Fig IP network, perform RTP session.

[0035] 例如,A,B两个参与者已经开始RTP会话,相互之间在发送RTP报文,SSRC没有冲突;这时C加入进来,C开始向A和B发送RTP数据,C的SSRC与A的SSRC冲突。 [0035] For example, A, B two players have started RTP sessions between each other not send RTP packets, SSRC conflict; then joined C, C starts sending RTP data to the A and B, C and the SSRC A conflict of SSRC. 下面分别介绍A,B, C这三个参与者的动作。 Introduced action A, B, C of the three participants below.

[0036] A =A收到C发送的RTP报文,发现C的SSRC与自己的一样,发生了冲突,这时,A立刻将自己的SSRC更换掉(自己随机产生一个新的SSRC标识),不过,重要的是,A的RTP报文的发送不受SSRC更换的影响,只是发送出去的RTP报文中的SSRC字段在某一个时刻发生了变化。 [0036] A = A C received RTP packets sent and found the SSRC C with their same clash, this time, A immediately replace the lost their SSRC (own randomly generates a new SSRC identifier) However, it is important, RTP packets are not sent SSRC A replacement effect, just send out packets in the RTP SSRC field in one moment changed.

[0037] C =C将收到A的RTP报文,发现A的SSRC与自己的一样,发生了冲突,这时,C立刻将自己的SSRC更换掉,RTP报文的发送保持不变。 [0037] C = C would receive A's RTP packets with their SSRC found A's, like, clashed, this time, C will own SSRC immediately replace the lost, RTP packets are sent unchanged.

[0038] B =B收到C发送的RTP报文后,发现C的SSRC与A的SSRC —样,发生了冲突,这时,B直接丢弃C的RTP报文,继续接收A的RTP报文。 After the [0038] B = B C received RTP packets sent and found the SSRC SSRC C and A's - like, clashed, then, B directly discards C of RTP packets, continues to receive A's RTP packets .

[0039] B要做到这一点,可以通过分析^^仙! [0039] B To do this, you can analyze ^^ cents! ! ⑶Number进行判断,虽然A发出的RTP 报文在某一个时刻SSRC发生了变化,但是kquence Number还是保持连续的,B通过分析Sequence Number的连续性来得到A发出的RTP报文,丢弃C发来的RTP报文。 ⑶Number judge, although A RTP packets sent at a certain moment SSRC has changed, but still maintain a continuous kquence Number, B Sequence Number by analyzing the continuity to get RTP packets sent by A discarded C sent RTP packets.

[0040] A和C都更换了SSRC之后,这个RTP会话就不再有SSRC冲突,A和B也就开始正常接收C的RTP报文了。 After the [0040] A and C are replaced with the SSRC, the SSRC RTP session there is no longer a conflict, A and B will start the normal reception of RTP packets of C.

[0041] 上述的实施例是最初建立RTP会话的参与者有两个的情形,本发明同样适用于最初参与者有三个或三个以上的情形,例如最初已建立会话的还有一个参与者D,则当A和新加入会话的C的同步源标识出现冲突时,参与者D的处理动作与上述实施例中的B的动作相同。 [0041] The embodiment described above is initially established RTP session participants two situations, the present invention is equally applicable to the initial three or more participants in the case, for example, there is initially established a session participant D , then when A synchronization source identification and newly added session C of conflict, participants D processing operation of the above embodiment B of the same action.

[0042] 综上所述,通过本发明的方法可以很快消除RTP会话中的SSRC冲突,并且通过对Sequence Number的判断,暂时丢弃新加入的参与者,保证了在SSRC发生冲突的时候已建立连接的语音不受影响,在发生冲突的SSRC均立即更新之后,只需短暂的时间就可以恢复通话。 [0042] In summary, by the method of the present invention can quickly eliminate conflict SSRC RTP session, and by the judgment of the Sequence Number, temporarily discard the newly added participants, to ensure that the conflict when the SSRC has been established Voice connections are not affected, in the aftermath of conflict SSRC are immediately updated, only a short time to be able to resume the call.

[0043] 需要注意的是,本发明有一个潜在默认条件,即在SSRC发生冲突的时候,SequenceNumber不能冲突。 [0043] It should be noted that the present invention has a potential default, when that is in conflict SSRC, SequenceNumber not conflict. 事实上,由于SSRC标识是32位随机数,其本身发生冲突的概率已经很小,Sequence Number的初始值也是16位的随机数,因此,SSRC标识与kquence Number两者同时发生冲突的概率可以认为是0。 In fact, since the SSRC identification is a 32-bit random number, the probability of conflict in itself is very small, the initial value of the Sequence Number is a 16-bit random number, therefore, the probability of SSRC identifies both kquence Number conflict can be considered It is 0.

Patent Citations
Cited PatentFiling datePublication dateApplicantTitle
CN1885857A20 Jun 200527 Dec 2006华为技术有限公司Method for recognizing RTP media stream in network
CN1996897A28 Dec 200511 Jul 2007中兴通讯股份有限公司A method for real time detection of the network transfer delay in the RTP
US200401655275 Dec 200326 Aug 2004Xiaoyuan GuControl traffic compression method
Classifications
International ClassificationH04L29/06, H04L29/08, H04M7/00
Legal Events
DateCodeEventDescription
20 Feb 2008C06Publication
30 Dec 2009C10Request of examination as to substance
25 May 2011C14Granted