CN1750505B - Sender Identification Method Based on Real-time Transport Protocol - Google Patents

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

Sender Identification Method Based on Real-time Transport Protocol

technical field

The invention relates to the technical field of real-time communication, in particular to a method for identifying a sender based on a real-time transport protocol.

Background technique

With the increasing demand for real-time communication services, some advanced real-time communication technologies, such as Voice over IP (VoIP, Voice over IP), have been more and more widely used. Among them, VoIP technology refers to a communication technology that converts voice information into digital voice packets of a certain length, and uses store and forward to exchange and transmit in the form of data packets. In addition to general point-to-point voice communication or video communication, VoIP service can also be applied to point-to-multipoint conference call service, conference TV service and push-to-talk (POC, Push To Talk Over Cellular) based on cellular network Conversational business, etc.

At present, in the VoIP communication system, voice signal flow or video media flow is exchanged by encapsulating data packets through Real-time Transport Protocol (RTP, Real-time Transport Protocol) and Real-time Transport Control Protocol (RTCP, Real-time Transport Control Protocol). Transmission. Among them, RTP and RTCP are protocols formulated by the Internet Engineering Task Force (IETF, Internet Engineering Task Force) for transmitting real-time information (including voice, video, etc.) The transmitted data packet encapsulation method, error control and quality of service (QoS, Quality of Service) detection method and other attributes.

Please refer to Fig. 1, which is a schematic diagram of the packet header format of an existing packet conforming to the Real-time Transport Protocol (RTP); wherein the RTP protocol stipulates that the synchronization source identifier (SSRC, Synchronization Source identifier) in the packet header is used to identify the signal flow or The sender of the media stream. Among them, in the point-to-point communication mode, since the only source of the RTP data packet received by the receiver is the other party who is currently talking with itself, the requirement for identifying the sender by the SSRC identifier is not very strict. However, in the multipoint communication mode, the RTP data packets received by the receiver may be sent by different senders. For example, in a VoIP conference call system, users may hear different people speaking at the same time, while in RTP In the protocol, the SSRC is randomly selected by the user during the initial session establishment process, so that the user who receives the RTP data packet cannot judge which sender sent the data packet according to the SSRC identifier, that is, the receiver can only Hear who is speaking directly based on the voice. If the receiver is not very familiar with the participants, they will not know who is speaking at all.

In the RTP protocol, the original intention of the SSRC logo is to identify the sender of the information, but due to the current random selection mechanism of the SSRC logo, the receiver of the RTP data packet cannot know the identity of the information sender through the SSRC logo. Communication has brought great inconvenience. Especially in a multi-person video conference system, when someone is speaking, other users can only identify who is currently speaking through the voice information of the speaker, and the user terminal cannot automatically display who is speaking.

Contents of the invention

The technical problem to be solved by the present invention is to provide a sender identification method based on the real-time transport protocol, which can use the synchronization source identification to identify the information sender in the multipoint communication system.

In order to solve the above problems, the present invention proposes a sender identification method based on real-time transport protocol, comprising steps:

(1) During the initial session establishment process, the session initiation user sends session establishment request information containing the user IDs of other participating users to the network side device, and the network side device responds to the session establishment request information and assigns synchronization Source ID, feed back the synchronization source ID allocated to the session initiating user and the user IDs and synchronization source IDs of other participating users to the session initiating user through the session establishment response information, and send the session initiating user's user ID, synchronization source ID, and other The user ID and synchronization source ID of each participating user are sent to each other participating user through the session establishment request message;

(2) The user terminal of each participating user stores the corresponding relationship between the user identification and the synchronization source identification of each participating user respectively;

(3) During the session, the sender user sends a real-time transport protocol packet with its own synchronization source identifier encapsulated in the packet header to other participating users;

(4) The user terminals of other participating users query the corresponding relationships stored by themselves according to the synchronization source identifier encapsulated in the header of the received real-time transport protocol data packet, and determine the corresponding user identifier to identify the sender user.

Wherein in the voice communication system based on the Internet Protocol, the network side device is a voice server based on the Internet Protocol. In the push-to-talk communication system based on the cellular network, the network side device is a push-to-talk server based on the cellular network.

Wherein the user identifier is a phone number or a user name.

In the sender identification method based on the real-time transport protocol of the present invention, during the initial session establishment process, the network side equipment assigns the synchronization source identification (hereinafter referred to as SSRC identification) to each participating user respectively, and the user terminal of each participating user stores the information of each participating user respectively. The corresponding relationship between the user ID of the user and the SSRC ID. In this way, the sender user who sends the real-time transport protocol data packet (hereinafter referred to as the RTP data packet) can encapsulate the SSRC ID assigned to itself in the header of the RTP data packet, and receive Other participating users who have received the RTP data packet sent by the sender user can query the corresponding relationships stored by themselves according to the SSRC identifier encapsulated in the header of the RTP data packet, and determine the corresponding user identifier to identify the sender user, thereby realizing the use of SSRC The purpose of identifying the sender of the RTP information by the identification is to solve the technical problem that the participating users cannot identify the sender of the RTP information in the multipoint communication system.

Description of drawings

Fig. 1 is a schematic diagram of the header format of an existing packet conforming to the Real-time Transport Protocol (RTP);

Fig. 2 is the flow chart of the main realization principle of the sender identification method based on the real-time transport protocol of the present invention;

Fig. 3 is a schematic diagram of the processing process of an embodiment of the method for identifying the sender based on the real-time transport protocol of the present invention.

Detailed ways

The design idea of the method for identifying the sender based on the real-time transport protocol of the present invention is to identify the sender of the RTP information through the SSRC identifier encapsulated in the header of the RTP data packet. That is, during the initial session establishment process, the network side equipment (such as the VoIP voice server in the VoIP communication system, the POC server in the POC communication system, etc.) assigns an SSRC identifier to each user participating in the session, so as to establish the SSRC identifier and The corresponding relationship of user identifiers (such as telephone numbers, user names, etc.), and at the same time, this corresponding relationship is sent to the user terminal of each user participating in the session for storage. In this way, during the session, the users participating in the session can directly query each corresponding relationship stored by themselves according to the information in the header SSRC identification field in the received RTP data packet, and determine the corresponding user ID to identify the sender of the RTP information.

The specific implementation of the method for identifying the sender based on the real-time transport protocol of the present invention will be described in detail below in conjunction with each drawing. Please refer to Fig. 2, this figure is the main realization principle flowchart of the sender identification method based on real-time transport protocol of the present invention; Its main realization process comprises:

Step S10, during the initial session establishment process, the network side equipment assigns SSRC identifiers to each participating user; the main allocation process is as follows:

The session initiating user sends session establishment request information including the user IDs of other participating users to the network side device;

The network side device responds to the session establishment request information sent by the session initiation user, and assigns an SSRC identifier to each participating user; and

Feedback the SSRC identifier allocated to the session initiating user and the user identifiers and SSRC identifiers of other participating users to the session initiating user through the session establishment response information; and

The user ID and SSRC ID of the session initiating user, and the user ID and SSRC ID of each other participating user are sent to each of the other participating users through the session establishment request message.

Wherein the user side equipment mentioned above is in the voice communication system (VoIP) based on the Internet protocol, and the described network side equipment is the voice server (i.e. the VoIP server) based on the Internet protocol; In a communication system (POC), the network side device is a push-to-talk server based on a cellular network (ie, a POC server).

Wherein the above-mentioned user identification can be a phone number or a user name.

Step S20, the user terminal of each participating user stores the corresponding relationship between the user identification and the SSRC identification of each participating user;

Step S30, during the session, the sender user sends an RTP packet with its own SSRC identifier encapsulated in the packet header to other participating users;

Step S40, the user terminals of other participating users receive the RTP data packet sent by the sender user, and query each corresponding relationship stored by themselves according to the SSRC identifier encapsulated in the header of the received RTP data packet, so as to determine the corresponding user identifier. Identifies the sending user.

Therefore, through the above processing procedure, the purpose of identifying the sender of the RTP information by using the SSRC identifier can be achieved.

Please refer to Fig. 3, which is a schematic diagram of the processing process of an embodiment of the sender identification method based on the real-time transport protocol of the present invention; here the implementation principle of the present invention is described in detail with the processing process of this embodiment, specifically as follows:

Step A, during the initial session establishment process, user 1 as the session initiating user sends session establishment request information to the network side device, wherein the session establishment request information sent includes the user identification of user 2 and the user identification of user 3;

Step B, the network side device receives the session establishment request information sent by user 1, and assigns SSRC identifiers to user 1, user 2 and user 3 respectively;

Step C, the network side device sends session establishment request information to user 2, the session establishment request information includes the SSRC allocated for user 2, and the user identification and SSRC identification of user 1, the user identification and SSRC identification of user 3);

Step D, after the user terminal of user 2 receives the session establishment request information, store the corresponding relationship between the user identification of each user and the SSRC;

Step E, the network side device sends session establishment request information to user 3, the session establishment request information includes the SSRC allocated for user 3, and the user identification and SSRC identification of user 1, the user identification and SSRC identification of user 2);

Step F, after the user terminal of user 3 receives the session establishment request information, store the corresponding relationship between the user identification of each user and the SSRC;

Step G, the network side device feeds back session establishment response information to user 1, the session establishment response information includes the SSRC allocated for user 1, the user identification and SSRC identification of user 2, the user identification and SSRC identification of user 3);

Step H, after the user terminal of user 1 receives the session establishment request information, store the corresponding relationship between the user identification of each user and the SSRC;

Step 1, user 1 speaks, and the user terminal of user 1 encapsulates the voice information of user 1 into an RTP voice data packet, and sends it to the network side device, wherein the header of the RTP voice data packet sent is encapsulated with the SSRC identification of user 1 ;

Step J, the network side device forwards the RTP voice data packet to the participating user 2;

Step K, the network side device forwards the RTP voice data packet to the participating users 3;

Step L, the user terminal of user 2 queries the respective correspondences stored by itself according to the SSRC identifier of user 1 encapsulated in the header of the received RTP voice data packet, determines the user identifier of user 1, and displays the user identifier of user 1 , so that user 2 knows that user 1 is making a speech through the user identification information displayed by his own user terminal;

Step M, the user terminal of user 3 queries the respective correspondences stored by itself according to the SSRC identification of user 1 encapsulated in the header of the received RTP voice data packet, determines the user identification of user 1, and displays the user identification of user 1 , so that user 3 can know that user 1 is making a speech through the user identification information displayed on his own user terminal.

Step N, user 3 speaks, and the user terminal of user 3 encapsulates the voice information of user 3 into an RTP voice data packet, and sends it to the network side device, wherein the packet header of the sent RTP voice data packet is encapsulated with the SSRC identifier of user 3 ;

Step 0, the network side device forwards the RTP voice data packet to the participating user 1;

Step P, the network side device forwards the RTP voice data packet to the participating user 2;

Step Q, the user terminal of user 1 queries the respective correspondences stored by itself according to the SSRC identification of user 3 encapsulated in the header of the received RTP voice data packet, determines the user identification of user 3, and displays the user identification of user 3 , so that user 1 knows that user 3 is making a speech through the user identification information displayed by his own user terminal;

Step R, the user terminal of user 2 queries the corresponding relationships stored by itself according to the SSRC identifier of user 3 encapsulated in the header of the received RTP voice data packet, determines the user identifier of user 3, and displays the user identifier of user 3 , so that user 2 can know that user 3 is making a speech through the user identification information displayed on his own user terminal.

The above process only takes the three-party communication as an example to illustrate the specific implementation of the sender identification method based on the real-time transport protocol of the present invention, and the process of applying the method of the present invention to multi-party communication is similar to the process of the above-mentioned three-party communication; and The above process only illustrates a feasible way of allocating SSRC identifiers, and may not be limited to the above processing process in the actual application process.

The above is only a preferred embodiment of the present invention, it should be pointed out that, for those of ordinary skill in the art, without departing from the principle of the present invention, some improvements and modifications can also be made, and these improvements and modifications can also be made. It should be regarded as the 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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