CN1750505B - Method for identifying sender based on real-time transmission protocol - Google Patents
Method for identifying sender based on real-time transmission protocol Download PDFInfo
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- CN1750505B CN1750505B CN2004100780175A CN200410078017A CN1750505B CN 1750505 B CN1750505 B CN 1750505B CN 2004100780175 A CN2004100780175 A CN 2004100780175A CN 200410078017 A CN200410078017 A CN 200410078017A CN 1750505 B CN1750505 B CN 1750505B
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Abstract
The invention discloses a sender identification method based on a real-time transport protocol, which comprises the steps that in the process of establishing an initial session, a network side device distributes a synchronous source identification for each participated user; the user terminal of each participating user respectively stores the corresponding relation between the user identification of each participating user and the synchronous source identification; in the session process, a sender user sends a real-time transmission protocol data packet with a header encapsulated with a self synchronization source identifier to other participating users; and the user terminals of other participating users inquire the corresponding relations stored by the user terminals according to the synchronous source identification encapsulated in the packet header of the received real-time transmission protocol data packet, and determine the corresponding user identification to identify the user of the sender. The invention can realize the identification of the RTP information sender by using the SSRC identification.
Description
Technical field
The present invention relates to the real time communication technical field, especially relate to a kind of transmitting side marking method based on RTP.
Background technology
Along with the continuous growth of real time communication business demand, the real time communication technology that some are advanced has obtained application more and more widely as ip voice business (VoIP, Voice over IP) etc.Wherein voip technology is meant the digitize voice bag that voice messaging is converted into certain-length, the communication technology that adopts mode that storage transmits to exchange and transmit with the form of packet.The VoIP business is except realizing general point-to-point voice communication or video communication, can also be applied to some the conference telephone service, video conferencing service of multiple spot and based on press-and-talk (POC, the Push To Talk Over Cellular) session service of Cellular Networks etc.
At present, in the VoIP communication system, sound signal stream or video media stream are by RTP (RTP, Real-time Transport Protocol) and RTCP Real-time Transport Control Protocol (RTCP, Real-time Transport Control Protocol) come encapsulated data packet to exchange transmission.Wherein RTP and RTCP are Internet engineering duty group (IETF, Internet Engineering Task Force) being used for of formulating transmitted the real time information agreement of (comprising information such as voice, video) on IP network, RTP and RTCP have stipulated the attributes such as packet encapsulation mode, error control and service quality (QoS, Quality ofService) detection mode that real time information is transmitted on IP network.
See also Fig. 1, this figure is the existing package head format schematic diagram that meets the packet of real-time transport protocol (rtp); Wherein the Synchronization Source in the Real-time Transport Protocol specified data bag packet header (SSRC, Synchronization Source identifier) is used for the transmit leg of id signal stream or Media Stream.Wherein in the point to point link mode and since unique source of the RTP packet that the recipient receives be current with the opposing party of self conversation, so be not very strict to requirement by SSRC mark for marking transmit leg.But in the multi-point mode, the RTP packet that the recipient receives may be to be sended over by different transmit legs, for example in the VoIP conference telephone system, the user may hear simultaneously that different people is in speech, and in Real-time Transport Protocol, SSRC sets up in the process at initial session and is selected at random by user self, so make the user who receives the RTP packet can not identify that to determine the end be the packet which transmit leg is sent according to SSRC, be that the recipient can only be directly recognize according to voice be who is speaking, if the recipient is not very familiar to the conference participation person, just can not know that the other side is who speaks at all.
In Real-time Transport Protocol, the original idea of SSRC sign is the transmit leg of identification information, but because the present choice mechanism at random of SSRC sign has caused the recipient of RTP packet to identify the identity of learning information transmitter by SSRC, this has brought very big inconvenience in multi-point.Especially in the TV conference system that many people participate in, when making a speech, it is who is making a speech that other users can only discern current by this spokesman's voice messaging as the people, and user terminal can't show it is who is in speech automatically.
Summary of the invention
The technical problem to be solved in the present invention is to propose a kind of transmitting side marking method based on RTP that can use Synchronization Source to come the identification information transmit leg in multipoint communication system.
For addressing the above problem, the present invention proposes a kind of transmitting side marking method based on RTP, comprise step:
(1) sets up in the process at initial session, session initiation user sends the session of the user ID that includes other each participant users and sets up solicited message to network equipment, the network equipment response session is set up solicited message, be respectively each participant user and distribute Synchronization Source, initiate user's Synchronization Source and other each participant users' user ID and Synchronization Source and set up response message by session and feed back to session and initiate the user distributing to session, session is initiated user's user ID and Synchronization Source, and other each participant users' user ID and Synchronization Source are set up solicited message by session and are sent to other each participant users respectively;
(2) each participant user's user terminal is stored each participant user's user ID and the corresponding relation between the Synchronization Source respectively;
(3) in conversation procedure, the transmit leg user sends packet header and is packaged with the RTP packet of self Synchronization Source to other participants user;
(4) each corresponding relation of the Synchronization Source inquiry that encapsulates in the packet header of other participants user's user terminal according to the RTP packet that receives self storage is determined relative users mark for marking transmit leg user.
Wherein in the voice communication system based on Internet protocol, described network equipment is the voice server based on Internet protocol.In the press-and-talk communication system based on Cellular Networks, described network equipment is the press-and-talk server based on Cellular Networks.
Wherein said user ID is telephone number or user name.
The transmitting side marking method that the present invention is based on RTP is by setting up in the process at initial session, be respectively each participant user by network equipment and distribute Synchronization Source (calling the SSRC sign in the following text), each participant user's user terminal is stored each participant user's user ID and the corresponding relation between the SSRC sign respectively, like this, the transmit leg user who sends RTP packet (calling the RTP packet in the following text) just can encapsulate the SSRC sign that self is assigned in the packet header of RTP packet, each corresponding relation that the SSRC sign inquiry that other participants user who receives the RTP packet that this transmit leg user sends encapsulates in just can the packet header according to the RTP packet self is stored, determine relative users mark for marking transmit leg user, thereby realized using the purpose of SSRC mark for marking RTP information sender, solved the technical problem that participant user in the multipoint communication system can't discern the RTP information transmitter.
Description of drawings
Fig. 1 is the existing package head format schematic diagram that meets the packet of real-time transport protocol (rtp);
Fig. 2 is the main realization principle flow chart that the present invention is based on the transmitting side marking method of RTP;
Fig. 3 is the embodiment processing procedure schematic diagram at the transmitting side marking method that the present invention is based on RTP.
Embodiment
The design philosophy that the present invention is based on the transmitting side marking method of RTP is: the transmit leg of the SSRC mark for marking RTP information by RTP packet packet header encapsulation.Promptly set up in the process at initial session, network equipment is (as the VoIP voice server in the VoIP communication system, POC server in the POC communication system etc.) distribute the SSRC sign for each user who participates in session, with the SSRC sign of setting up each user and the corresponding relation of user ID (as telephone number, user name etc.), the user terminal that this corresponding relation is sent to each user of participation session is stored simultaneously.Like this, in conversation procedure, the user who participates in session just can directly inquire about each corresponding relation of self storing according to the information of the RTP packet middle wrapping head SSRC identification field of receiving, determines that corresponding user ID identifies the transmit leg of RTP information.
Be explained in detail below in conjunction with the embodiment of each accompanying drawing the transmitting side marking method that the present invention is based on RTP.See also Fig. 2, this figure is the main realization principle flow chart that the present invention is based on the transmitting side marking method of RTP; Its main implementation procedure comprises:
Step S10 sets up in the process at initial session, is respectively each participant user by network equipment and distributes the SSRC sign; Its main assigning process is as follows:
Session initiation user sends the session of the user ID that includes other each participant users and sets up solicited message to network equipment;
Solicited message is set up in the session that this network equipment response session initiation user sends, and is respectively each participant user and distributes the SSRC sign; And
Initiate user's SSRC sign and other each participant users' user ID and SSRC sign and set up response message by session and feed back to session and initiate the user distributing to session; And
With session initiate user's user ID and SSRC sign, and other each participant users' user ID and SSRC sign set up solicited message by session and send to other each participant users respectively.
The wherein above-mentioned user side equipment of mentioning is in the voice communication system (VoIP) based on Internet protocol, and described network equipment is the voice server (being the VoIP server) based on Internet protocol; And in the press-and-talk communication system (POC) based on Cellular Networks, described network equipment is the press-and-talk server (being POC server) based on Cellular Networks.
The wherein above-mentioned user ID of mentioning can or be user name for telephone number.
Step S20, each participant user's user terminal store each participant user's user ID and the corresponding relation between the SSRC sign respectively;
Step S30, in conversation procedure, the transmit leg user sends packet header and is packaged with the RTP packet of self SSRC sign to other participants user;
Step S40, other participants user's user terminal receives the RTP packet that the transmit leg user sends, the SSRC that encapsulates in the packet header according to the RTP packet that receives identifies and inquires about each corresponding relation of self storing, to determine that corresponding user ID identifies the transmit leg user.
Thereby, just can realize by using the purpose of SSRC mark for marking RTP information sender by above-mentioned processing procedure.
See also Fig. 3, this figure is the embodiment processing procedure schematic diagram at the transmitting side marking method that the present invention is based on RTP; Here the processing procedure with this embodiment describes realization principle of the present invention in detail, and is specific as follows:
Steps A is set up in the process at initial session, and the user 1 who initiates the user as session sends session and sets up solicited message to network equipment, and wherein the user ID that includes user 2 in the solicited message and user 3 user ID are set up in the session of Fa Songing;
Step B, network equipment receive the session that user 1 sends and set up solicited message, are respectively user 1, user 2 and user 3 and distribute the SSRC sign;
Step C, network equipment sends session to user 2 and sets up solicited message, this session is set up and is comprised the SSRC that promising user 2 distributes in the solicited message, and user 1 user ID and SSRC sign, user 3 user ID and SSRC sign);
Step D, user 2 user terminal receive after session sets up solicited message, store each user's user ID and the corresponding relation between the SSRC;
Step e, network equipment sends session to user 3 and sets up solicited message, this session is set up and is comprised the SSRC that promising user 3 distributes in the solicited message, and user 1 user ID and SSRC sign, user 2 user ID and SSRC sign);
Step F, user 3 user terminal receive after session sets up solicited message, store each user's user ID and the corresponding relation between the SSRC;
Step G, network equipment is set up response message to user's 1 feeding back conversation, this session is set up and is comprised the SSRC that promising user 1 distributes in the response message, and user 2 user ID and SSRC sign, user 3 user ID and SSRC sign);
Step H, user 1 user terminal receive after session sets up solicited message, store each user's user ID and the corresponding relation between the SSRC;
Step I, user 1 makes a speech, and user 1 user terminal is handled user 1 voice messaging and is packaged into the RTP VoP, and sends to network equipment, and wherein the packet header of the RTP VoP of Fa Songing is packaged with user 1 SSRC sign;
Step J, network equipment transmit the user 2 that this RTP VoP is given participant;
Step K, network equipment are transmitted the user 3 that this RTP VoP is given participant;
Step L, the user's 1 who encapsulates in the packet header of user 2 user terminal according to the RTP VoP that receives SSRC sign, inquire about each corresponding relation of self storing, determine user 1 user ID, and user 1 user ID shown, so that user 2 learns it is that user 1 is making a speech by the user totem information of self user terminal displays;
Step M, the user's 1 who encapsulates in the packet header of user 3 user terminal according to the RTP VoP that receives SSRC sign, inquire about each corresponding relation of self storing, determine user 1 user ID, and user 1 user ID shown, so that user 3 learns it is that user 1 is making a speech by the user totem information of self user terminal displays.
Step N, user 3 makes a speech, and user 3 user terminal is handled user 3 voice messaging and is packaged into the RTP VoP, and sends to network equipment, and wherein the packet header of the RTP VoP of Fa Songing is packaged with user 3 SSRC sign;
Step O, network equipment transmit the user 1 that this RTP VoP is given participant;
Step P, network equipment transmit the user 2 that this RTP VoP is given participant;
Step Q, the user's 3 who encapsulates in the packet header of user 1 user terminal according to the RTP VoP that receives SSRC sign, inquire about each corresponding relation of self storing, determine user 3 user ID, and user 3 user ID shown, so that user 1 learns it is that user 3 is making a speech by the user totem information of self user terminal displays;
Step R, the user's 3 who encapsulates in the packet header of user 2 user terminal according to the RTP VoP that receives SSRC sign, inquire about each corresponding relation of self storing, determine user 3 user ID, and user 3 user ID shown, so that user 2 learns it is that user 3 is making a speech by the user totem information of self user terminal displays.
Above flow process only to be example with the Three Party Communication be illustrated the concrete performance of the transmitting side marking method that the present invention is based on RTP, and it is similar to use the process of the processing procedure of the method for the invention and above-mentioned Three Party Communication for multi-party communication; And top flow process has only illustrated the feasible pattern that a kind of SSRC sign is distributed, the handling process above can being not limited in actual application.
The above only is a preferred implementation of the present invention; should be pointed out that for those skilled in the art, under the prerequisite that does not break away from the principle of the invention; can also make some improvements and modifications, these improvements and modifications also should be considered as protection scope of the present invention.
Claims (4)
1. the transmitting side marking method based on RTP is characterized in that, comprises step:
(1) sets up in the process at initial session, session initiation user sends the session of the user ID that includes other each participant users and sets up solicited message to network equipment, the network equipment response session is set up solicited message, be respectively each participant user and distribute Synchronization Source, initiate user's Synchronization Source and other each participant users' user ID and Synchronization Source and set up response message by session and feed back to session and initiate the user distributing to session, and session is initiated user's user ID and Synchronization Source, and other each participant users' user ID and Synchronization Source are set up solicited message by session and are sent to other each participant users respectively;
(2) each participant user's user terminal is stored each participant user's user ID and the corresponding relation between the Synchronization Source respectively;
(3) in conversation procedure, the transmit leg user sends packet header and is packaged with the RTP packet of self Synchronization Source to other participants user;
(4) each corresponding relation of the Synchronization Source inquiry that encapsulates in the packet header of other participants user's user terminal according to the RTP packet that receives self storage is determined relative users mark for marking transmit leg user.
2. the transmitting side marking method based on RTP according to claim 1 is characterized in that, in the voice communication system based on Internet protocol, described network equipment is the voice server based on Internet protocol.
3. the transmitting side marking method based on RTP according to claim 1 is characterized in that, in the press-and-talk communication system based on Cellular Networks, described network equipment is the press-and-talk server based on Cellular Networks.
4. the transmitting side marking method based on RTP according to claim 1 is characterized in that, described user ID is telephone number or user name.
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| Application Number | Priority Date | Filing Date | Title |
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| CN2004100780175A CN1750505B (en) | 2004-09-16 | 2004-09-16 | Method for identifying sender based on real-time transmission protocol |
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| Application Number | Priority Date | Filing Date | Title |
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| CN2004100780175A CN1750505B (en) | 2004-09-16 | 2004-09-16 | Method for identifying sender based on real-time transmission protocol |
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| CN1750505A CN1750505A (en) | 2006-03-22 |
| CN1750505B true CN1750505B (en) | 2010-04-28 |
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| CN2004100780175A Expired - Lifetime CN1750505B (en) | 2004-09-16 | 2004-09-16 | Method for identifying sender based on real-time transmission protocol |
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Families Citing this family (10)
| Publication number | Priority date | Publication date | Assignee | Title |
|---|---|---|---|---|
| CN101291445B (en) * | 2007-04-20 | 2014-04-30 | 华为技术有限公司 | Service data manipulating method, device and system by session initiative protocol |
| CN101039325B (en) * | 2007-04-26 | 2010-12-29 | 中兴通讯股份有限公司 | Method for configuring real-time transmission protocol packet based on mixer |
| CN101127772B (en) * | 2007-09-25 | 2011-04-20 | 中兴通讯股份有限公司 | Method for distributed processing of real time transfer of protocol signaling |
| CN104079870B (en) * | 2013-03-29 | 2017-07-11 | 杭州海康威视数字技术股份有限公司 | The video frequency monitoring method and system of single channel multi-channel video audio |
| CN106534108A (en) * | 2016-10-27 | 2017-03-22 | 武汉船舶通信研究所 | Call party identification method, switch and command voice terminal under multi-party communication scene |
| CN110290130B (en) * | 2019-06-21 | 2020-09-01 | 京信通信系统（中国）有限公司 | VOLTE data transmission method and device, access network equipment and storage medium |
| CN112311503A (en) * | 2019-07-25 | 2021-02-02 | 华为技术有限公司 | Video data transmission method and device |
| CN111314319B (en) * | 2020-01-20 | 2022-06-10 | 维沃移动通信有限公司 | Data processing method, electronic equipment and server |
| CN112218252A (en) * | 2020-10-10 | 2021-01-12 | 海能达通信股份有限公司 | Identity information processing method and device |
| CN116074553B (en) * | 2023-01-28 | 2023-06-23 | 天翼云科技有限公司 | Video stream transmission method, device, electronic equipment and storage medium |
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|---|---|---|---|---|
| CN1309511A (en) * | 2000-02-14 | 2001-08-22 | 三洋电机株式会社 | Radio base station and radio telephone device |
| CN1461574A (en) * | 2001-04-17 | 2003-12-10 | 诺基亚公司 | One-to-one communication |
| CN1458788A (en) * | 2002-05-14 | 2003-11-26 | 华为技术有限公司 | Full network marking method for common dialing user |
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