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A method for solving synchronization source identity confliction in RTP session

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Publication number
CN101127712B
CN101127712B CN 200710143006 CN200710143006A CN101127712B CN 101127712 B CN101127712 B CN 101127712B CN 200710143006 CN200710143006 CN 200710143006 CN 200710143006 A CN200710143006 A CN 200710143006A CN 101127712 B CN101127712 B CN 101127712B
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participator
source
synchronous
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conflict
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CN 200710143006
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Chinese (zh)
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CN101127712A (en )
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周蕙菁
张新林
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中兴通讯股份有限公司
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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.

Description

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

技术领域 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 belongs to the VoIP (Voice over Internet Protocol, IP i I sound technology) technologies, and particularly relates to a case where the RTCP (RTP Contro 1 Protoco 1, RTP control protocol) is not activated, solving RTP (Real- Time transport protocol, real-time transport protocol) session conflicts method SSRC (synchronization source, synchronization source identifier).

背景技术 Background technique

[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-end real-time data transmission, 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, sequence number codes, time stamp, the transmission monitor. 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) protocol, using UDP multiplexing, packet check 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 comprises two parts: RTP (real-time data for transmission) and the RTCP (RTP for monitoring quality of service) information interaction session. 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 part is completed package, serial number management, the time stamp coding, synchronization source identifier function. RTCP周期地向RTP会话的各方发送控制报文。 RTCP periodically transmits to the parties RTP session control packets. 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 is to implement: providing a data transmission quality feedback; RTP as the transport protocol which is to provide flow control, congestion control part, the feedback control can be used for adaptive coding, for monitoring local or remote error for third-party IP multicast for network monitoring available to use. 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 packet into SR (Sender Report), RR (receiver report), SDES (Source Description), BYE (session end), APP (application-specific message) five types.

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

[0005] 版本号V(Version) :2位,代表RTP的版本号; [0005] The version number V (Version): 2 bits representative 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 identification PO ^ dding): 1 bit when P = 1 means comprises one or more padding bytes to 32-bit alignment;

[0007] 扩展位X(EXtension) :1位,表示是否包括扩展头部; [0007] The extension bit X (EXtension): 1 bit, indicates whether the extension header comprising;

[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 bits represents CSRC count (the Count CSRC);

[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 important events in the packet stream marked as a frame boundary;

[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 identify the format of the RTP payload, which determines how to load the application decoder;

[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] sequence number (kquence Number): 16 bits for detecting when sorting RTP packet loss and packet reordering reference, such as a sender sends a RTP SSRC each packet, the SequenceNumber increase the initial value of the random Sequence Number generating data packet sequence receiver can detect the packet loss by the domain and restore;

[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] stamp (Timestamp): 32 bit sampling instant of the first byte of the record data packet;

[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] The synchronization source identifier (SSRC): 32 bits identify a source of the RTP data stream, generating the random SSRC, so that, when it is possible to have a plurality of participants, 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] Contributing Source identifier (CSRC, Contributing Source): 0~15 term, the contribution of each source 32 to the RTP payload data packet identification.

[0015] RFC3550是RTP协议的标准。 [0015] RFC3550 RTP is a standard 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, to all participants unique SSRC field; if a RTP session participant found SSRC collisions (i.e., a RTP session, the participants SSRC not unique), by sending RTCP BYE message, exit the session, and then rebuild a SSRC identifier, join the conversation;

[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, when the multi-party session, a participant if a discovery session RTP SSRC collisions other two participants, as will be distinguished by determining whether the network packets or transport layer addresses in RTCP SDES content determination, SSRC until no longer 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 a RTP session, a new SSRC other participants are considered as conditions effective data stream is: received at least two consecutive RTP packets sent from the SSRC.

[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 is not necessary to enable. 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。 RTCP can not enable the 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 conflict SSRC, how to solve, the agreement did not specify.

发明内容 SUMMARY

[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 present invention is to solve the technical problem is to provide a method in the case of RTCP is not activated, solving session RTP SSRC conflict.

[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 a method for identifying synchronization source RTP session conflict resolution, after establishing an RTP session, if a participant in the session a synchronization source identification found their new added to the synchronization source identifier of a session conflict two participants, then the updated immediately own synchronization source identifier, and continues to send packets; when the two participants found their synchronization source identifier of the participant is also a conflict immediately update its synchronization source identifier, and continues to send the message; the other participants in the session participants found two synchronized source identification conflict with a participant, the participant packet discards II, continue 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] Furthermore, the above method may also have the following characteristics: a participant in the updated own synchronization source identification, in accordance with conventional frequency continues to send packets, the other participants in the session by analyzing the sequence number in the received packet to obtain a packet sent by the participant.

[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] Furthermore, the above method may also have the following characteristics: after the participant twelve synchronization source identifier are immediately updated with the participant, the other participants and a participant in the session, i.e. normally receiving the participant sending two 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] Furthermore, the above method may also have the following characteristics: after the two participants join the session, starts sending packets, said one participant in the participant receives the second transmission message hereinafter, we found their synchronization source identifier conflicts with the actor two.

[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] Furthermore, the above method may have the following features: the two participants in the participant after receiving a packet transmitted, find their synchronization source identifier and a participant of the 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] Furthermore, the above method may have the following features: the first RTP session participant to have 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] Furthermore, the above method may also have the following characteristics: updating the synchronization source identification is done by 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] RTP SSRC collisions can quickly eliminate the session by the method of the present invention, and by analyzing the packets Sequence Number to ensure that the voice connection has been established at the time SSRC conflict affected, in the event after SSRC conflict are updated immediately, just a short time it can 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 diagram of embodiments of the present invention will be more participants 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 [0031] The present invention solves SSRC conflict (not enabled RTCP) is: once a conflict is found, the participant's own SSRC identifier update immediately, and by determining the transmission frequency or sequence number field, as much as possible to ensure that RTP packets are 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 only real-time data transmission RTP itself does not provide any ability to guarantee real-time transmission of data and quality of service, therefore, the present invention is provided by the RTP sequence number in the receiving side is determined according to the information in the packet 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] Specific embodiments of the present invention will be described in further detail below in conjunction with the accompanying drawings and.

[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] Referring to FIG. 2, A, B, C are connected by three participants IP network, the 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 has two participants start the RTP session, between each other without sending RTP packet, SSRC conflict; joined case C, C to start sending RTP data 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。 The following describes the operation of A, B, C of the three participants.

[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 and C as its own, clashed, this time, A will immediately replace the lost their SSRC (own randomly generated a new SSRC identifier), However, it is important, RTP packet sent from a SSRC impact of replacement, just send out RTP packets SSRC field changes at a certain moment.

[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 will receive the RTP packets of A, A is found with their SSRC like clashed, this time, C own SSRC immediately replace the lost, RTP packet transmission remains 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 [0038] B = B C receives RTP packets transmitted, found SSRC SSRC C and A is - like, a conflict occurs, then, B discards RTP packets of C, A continues to receive the RTP packet .

[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 judgment, although RTP packet A sends a change at a certain moment the SSRC, but still maintain a continuous kquence Number, B to A sends an RTP packet obtained by analyzing the continuity of Sequence Number, discarding incoming C 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 [0040] A and C are replaced SSRC, the RTP session there is no longer SSRC conflict, A and B will start RTP packet of the normal receiving 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 first embodiment is established RTP session participant has two circumstances, the present invention is equally applicable to three or more participants initially or three situations, for example, there is initially established a session participant D , when the synchronization source identification a and the newly added session conflict of C, D and the processing operation of the participant B in the operation same as the above-described embodiments.

[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] As described above, can be quickly eliminated by the method according to the present invention SSRC collisions RTP session, and by the judgment of Sequence Number temporarily discarding the newly added participant to ensure that when the conflict has been established in the SSRC voice connections are not affected, after the SSRC conflict are updated immediately, just a short time you can 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] Note that, there is a potential default conditions of the present invention, i.e., when the conflict occurs 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 identifier 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, thus the probability of two simultaneous kquence Number SSRC identifies the conflict can be considered It is 0.

Claims (7)

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 solution RTP session synchronization source identifier conflict, wherein, after establishing an RTP session, if a participant in the session a synchronization source identification found their new participants join the session II synchronization source identifier conflict, the synchronization update their immediate source identification, and continues to send the message; the two participants can update its synchronization source identifier immediately of their synchronization source identifier and a participant of the conflict, and continues to send the message; the other participants in the session participants found two synchronization source identifier and a participant conflict, discards the packet two participants, the participants continue to receive a message.
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, wherein a participant in the updated own synchronization source identification, to continue transmitting packets according to established frequency, the other participants in the session by analyzing the sequence number in the received packet to obtain a packet sent by the participant.
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, wherein, after said twelve participants are updated immediately identifies the synchronization source and the participant, the other participants and a participant in the session, i.e. normally receiving the participant sending two 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 claim 1 or 2 or as claimed in claim 3, wherein said two participants join the session, starts sending packets, one participant in the participant receives the second transmission after the message, find their synchronization source identifier conflicts with the actor two.
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, characterized in that, the second is the participant after receiving a message sent by the participant, they found their synchronization source identifier and a participant of the 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。 The method as claimed in claim 5, characterized in that said first RTP session participant to have 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 identifier update is performed by the real-time transport protocol.
CN 200710143006 2007-08-20 2007-08-20 A method for solving synchronization source identity confliction in RTP session CN101127712B (en)

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CN1885857A (en) 2005-06-20 2006-12-27 华为技术有限公司 Method for recognizing RTP media stream in network
CN1996897A (en) 2005-12-28 2007-07-11 中兴通讯股份有限公司 A method for real time detection of the network transfer delay in the RTP

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US20040165527A1 (en) 2002-12-20 2004-08-26 Xiaoyuan Gu Control traffic compression method
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CN1996897A (en) 2005-12-28 2007-07-11 中兴通讯股份有限公司 A method for real time detection of the network transfer delay in the RTP

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