CN1838790A - PTT service realizing system and method based on VoIP technique - Google Patents

Abstract

The present invention provides a system and method for implementing a push-to-talk service based on VoIP technology, including a push-to-talk client, an online user server, and a broadband gateway. When the above-mentioned online server sends a call request, the online server judges the network attribute of the user in the call group according to the request, forwards the call signaling to the called end according to the attribute, and establishes a media channel between users to exchange media packets. For users, the online server forwards the call signaling, and directly establishes a media channel between users. For private network users, the online server forwards the call signaling, and conducts signaling negotiation with the broadband gateway, and establishes a media channel through the broadband gateway. The present invention can Adaptively distinguish different PTT users, and adopt different media processing schemes according to user attributes, which greatly improves the efficiency of PTT calls and reduces the load on system equipment.

Description

"Push to talk" service realization system and method based on VoIP technology

technical field

The invention relates to the field of instant messaging, in particular to a VoIP technology-based push-to-talk service realization system and method.

Background technique

In today's information age, people always hope to be able to talk immediately when they press a certain key, instead of dialing 11 digits first like public mobile phones, and waiting for several milliseconds or longer to make a call. You don’t know when you can talk, no matter how urgent you are, you can only wait patiently, and PTT (Push To T calling terminal lk), that is, "one-touch talk" service, can quickly establish a call and provide a similar trunking system for mobile phone users Communication services of "one-to-one" or "one-to-many" half-duplex calls.

PTT originated from the CDMA network in North America, and the PTT service based on CDMA 1x has attracted many users in a very short period of time. Later, major operators and manufacturers followed up one after another and launched various related PTT services. At present, this technology has been successfully applied to GSM and CDMA networks.

In the public mobile network, no matter it is GSM or CDMA, its PTT adopts all-digital VoIP technology, which is designed according to Session Initiation Protocol (SIP) and Multimedia Subsystem (IMS). This technology is based on packet data network, storage switching The principle is different from the direct transmission and exchange of the circuit in the cluster professional network.

The continuous development of broadband network technology and the widespread application of broadband network provide the possibility for the realization and large-scale use of the "one-touch" service based on VoIP technology. However, as we all know, the lack of IP addresses has caused the complexity of the network topology. Therefore, in the existing PTT implementation schemes based on VoIP technology, it is necessary to solve the problem of PTT calls between users in different LANs.

In order to solve the network traversal problem of private network users, usually, PTT systems based on VoIP technology need to add a system device to control and process media packets. However, how to effectively improve the efficiency of PTT calls and reduce the load on system equipment is one of the key problems to be solved in the PTT system based on VoIP technology.

Contents of the invention

The technical problem to be solved by the present invention is to provide a method for implementing a push-to-talk service based on VoIP technology, which can improve the efficiency of PTT calls while realizing PTT calls between users of different LANs, shorten the call setup time, and reduce the cost of system equipment. Load, improve the real-time performance of media transmission.

The present invention provides a PTT service implementation system based on VoIP technology, including a PTT client for initiating/accepting a PTT call request, and encoding and decoding media streams, and also includes:

The user online server is used to accept the registration of the client, record the connection attribute of the client, and forward the push-to-talk call request accordingly, and assist the client in the push-to-talk call in which only the public network client participates. directly establish media channels between them, and exchange media packets;

The broadband gateway device is connected to the user online server through an internal interface, and is used to perform signaling negotiation with the online server in a push-to-talk call participated by a private network client with a network address translation device, and acts as an intermediary for The client establishes a media channel and exchanges media packets.

The present invention further provides a method for realizing a push-to-talk service based on VoIP technology, comprising the following steps:

(1) The push-to-talk client registers with the user's online server;

(2) The PTT calling initiator sends a call request to the online server;

(3) The online server judges whether the network attributes of the users in the calling group belong to public network users or private network users according to the call request;

(4) The online server forwards the call signaling to the called end according to the network attribute, assists users in establishing media channels, and exchanges media packets,

Wherein, for the call that only public network users participate in, the call signaling is forwarded by the online server, and a media channel is directly established between users to exchange media packets;

For the call participated by the private network user with the network address translation function, the online server forwards the call signaling, and conducts signaling negotiation with the broadband gateway, establishes a media channel via the broadband gateway, and exchanges media packets.

In the existing network architecture, the present invention realizes a PTT service based on VoIP technology, can adaptively distinguish different PTT users, adopts different media processing schemes according to user attributes, greatly improves the efficiency of PTT calls, and further It shortens the call establishment time, effectively reduces the load of system equipment, and improves the real-time performance of media transmission.

Description of drawings

Fig. 1 is a schematic diagram of a system networking structure applying the present invention;

Fig. 2 is the realization method flowchart of the present invention;

Fig. 3 is the PTT call interface and process description of public network user "one-to-one" in the present invention;

Fig. 4 is the "one-to-one" PTT call interface and process description that private network users participate in in the present invention;

Fig. 5 is a "one-to-many" PTT call interface and process description in which private network users participate in the present invention.

Detailed ways

The present invention provides a system and method for implementing a "push to talk" service based on VoIP technology. The used equipment mainly includes: PTT client, user online server PS and broadband gateway equipment BGW. The PTT client can initiate or accept PTT call requests; the user online server is responsible for processing PTT call requests. The gateway determines the media forwarding criteria; the broadband gateway device does not participate in PTT calls between public network users, it plays an important role in the PTT call process with private network users, and the forwarding of media packets will be completed by the broadband gateway device. The present invention adopts a Client-Server (client-server) architecture configuration system, and the SIP (Session Initiation Protocol) protocol is used for communication between the PTT client and the online server, and an internal interface is used between the online server and the broadband gateway device. The client's request or response is encapsulated into the MESSAGE source language of the SIP message for transmission.

Both sides of the PTT call describe their own media according to the Session Description Protocol (SDP, Session Description Protocol), then encapsulate the SDP into a SIP-MESSAGE and send it to the online server, and the online server forwards the message. After receiving the other party's message, the calling parties extract and decode the other party's SDP description, further completing the media negotiation and the establishment of the media channel. For two public network users, the two parties directly exchange their SDP descriptions through the online server to establish the media channel of the PTT call; and for the private network users with the network address translation function, the PTT call signaling is also forwarded by the online server, but , the online server and the broadband gateway must also conduct signaling negotiation for establishing a media channel, and the broadband gateway establishes a media channel, formulates a media forwarding mechanism, and realizes the exchange of media packets between the two parties.

The present invention can adaptively distinguish different PTT users, and adopt different media processing schemes according to user attributes. When the PTT client registers with the online server, it needs to send its own connection attributes to the online server through the MESSAGE source language of the SIP protocol. The connection properties mainly include: the IP address of the computer where the client is located and the port number of the connection. Based on the information and the address and port number used by the PTT client to register with the online server, the online server can judge the network attribute of the client. When making a PTT call, it can be determined whether the users in the PTT call group are public network users or whether they are located in the same local area network.

In the present invention, the PTT call includes not only audio stream, but also video stream. Both audio and video encoding and decoding are done on the PTT client. Among them, the audio processing supports G.711, G.729 and G.723 standards, and the video codec adopts the H.263 standard.

The "public network user" mentioned in the present invention not only refers to the public network user on the Internet, but also includes the user located in the same local area network; "private network user" refers to the user who is not in the same local area network.

As shown in FIG. 1 , it is a schematic diagram of a system networking structure applying the present invention. This system is composed of PTT client, user online server and broadband gateway, wherein, the private network client has network address translation (NAT) equipment. The SIP protocol can be used for signaling interaction between the PTT client and the online server; after the PTT call is established, the media transmission between the clients and between the client and the broadband gateway can use the RTP/RTCP protocol; the online server and Internal self-defined interfaces can be used between broadband gateways. The PTT client can initiate or accept PTT call requests, and it is also responsible for the encoding and decoding of audio and video media streams; the user online server is responsible for the registration and registration of the PTT client, while recording the connection attributes of the PTT client, and completing the forwarding of the PTT call request ; The broadband gateway device is used for PTT calls participated by private network users, and it completes the control and distribution of media packets. PTT calls between public network users have nothing to do with broadband gateways.

As shown in Figure 2, it is the implementation method flowchart of the present invention, at first the push-to-talk client registers with the user online server (step 201); when the push-to-talk calling initiator sends a call request to the online server ( Step 202); The online server judges according to the call request that the network attribute of the user in the call group belongs to the public network user or has a private network user (step 203); then the online server forwards the call signal to the called terminal according to the network attribute order, and assist users to set up media channels and exchange media packets (step 204). For calls that only public network users participate in, the online server forwards the call signaling, directly establishes media channels between users, and exchanges media packets. The online server forwards the call signaling for the private network users with the network address translation function, and conducts signaling negotiation with the broadband gateway, establishes a media channel via the broadband gateway, and exchanges media packets.

In the present invention, since there is a big difference between the PTT call between public network users and the PTT call flow involving private network users, it is necessary to judge and distinguish the network attributes of each client user. This work can be done during the registration process of the PTT client with the online server.

A PTT client must first register and register with the online server. Registration and registration are done using the SIP protocol. After the registration is successful, the PTT client also needs to send its own connection attributes to the online server through the MESSAGE source language of the SIP protocol. The connection properties mainly include: the IP address of the computer where the client is located and the port number of the connection. At the same time, the online server can obtain the address and port number used by the PTT client to register with the online server according to the Socket API. By comparing the two, the network attribute of the client can be judged to determine whether the user is a private network user or a public network user.

As shown in Figure 3, for public network users, after PTT clients A and B register with the online server, A initiates a PTT call to B, and the process is as follows:

1. A describes its own media information in SDP, encapsulates the SDP into a SIP-MESSAGE message, and sends it to the online server;

2. The online server checks the status of B, if B is not idle, then directly sends a call failure message to A; otherwise, the online server forwards the PTT call request to B; and records the owner of the current speaking right (belonging to A) ;

3. After receiving the PTT call request, B opens the media channel, prepares to receive the media packet from A, and decodes and plays it; at the same time, encapsulates the PTT call response into a SIP-MESSAGE message and sends it back to the online server;

4. The online server sends B's call response message to A;

5. After receiving B's PTT call response, A starts to collect local media, and sends it to B after compressing and encoding;

6. After A finishes speaking, stop sending the media package, and send a message to the online server to release the right to speak, and the online server notifies B;

7. A and B send applications for the right to speak to the online server, and the online server allocates the right to speak on a first-come, first-served basis, and records the current owner of the right to speak;

8. Repeat steps 1 to 7;

9. After the call ends, the on-hook party sends an on-hook message to the online server, and the online server notifies the other party to hang up.

As shown in Figure 4, for a "one-to-one" PTT call involving private network users, after PTT clients A and B register with the online server, A initiates a PTT call to B, and the process is as follows:

1. A describes his media information with SDP, and encapsulates the SDP into a SIP-MESSAGE message, and then sends the message to the online server through the NAT (Network Address Translation) device of the local area network;

2. The online server checks the status of B. If B is not idle, it sends a call failure message to A; otherwise, the online server forwards the PTT call request to the NAT device of the LAN where B is located, and the NAT device sends it to A. B;

3. The online server sends PTT call information to the broadband gateway;

4. The broadband gateway records the PTT call information, and prepares to receive and forward the media packet from A;

5. After receiving the PTT call request, B opens the media channel, prepares to receive the media packet from A, and decodes and plays it; at the same time, encapsulates the PTT call response into a SIP-MESSAGE message, and sends it back to the online server through the NAT device;

6. B sends a UDP message to the broadband gateway, so as to open up the routing channel for the broadband gateway to forward the media to B;

7. The online server sends B's call response message to A through A's NAT device;

8. After receiving B's PTT call response, A starts to collect local media, and sends it to the broadband gateway through the NAT device after compression and encoding;

9. The broadband gateway forwards A's media packet to B;

10. After A finishes speaking, stop sending the media package, and send a message to the online server to release the right to speak, and the online server notifies B;

11. A and B send applications for the right to speak to the online server, and the online server allocates the right to speak on a first-come, first-served basis, and records the current owner of the right to speak;

12. Repeat steps 1 to 11;

13. After the call ends, the on-hook party sends an on-hook message to the online server, and the online server notifies the other party to hang up; at the same time, the online server sends a call-end message to the broadband gateway, and the broadband gateway deletes the call information.

As shown in Figure 5, the "one-to-many" PTT call process involving private network users, taking three parties as an example, after PTT clients A, B, and C register with the online server, A online server sends a PTT call request, requesting to hold a PTT call For a PTT call from A to B and C, the process is as follows:

1. A describes its own media information in SDP, encapsulates the SDP into a SIP-MESSAGE message, and then sends the message to the online server;

2. The online server checks the online status, and if the check is passed, the PTT call request is forwarded to B and C, and the owner of the current speaking right (belonging to A) is recorded at the same time;

3. The online server sends PTT call information to the broadband gateway;

4. The broadband gateway records the PTT call information, and prepares to receive A's media packet and forward it to B and C;

5. After B and C receive the PTT call request, they open the media channel, prepare to receive A's media from the broadband gateway, and decode and play it;

6. B and C send a call response message to the online server, and the online server forwards it to A;

7. B and C send UDP messages to the broadband network management, so as to open up the routing channel for the broadband gateway to forward media to B and C;

8. After receiving B's PTT call response, A opens the media channel, starts collecting local media, and sends it to the broadband gateway after compressing and encoding;

9. The broadband gateway forwards the media packet of A to B and C respectively;

101. After A finishes speaking, he sends a message to the online server to release the right to speak, and the online server notifies B and C; at the same time, the online server sends a message to the broadband gateway, and the broadband gateway stops the forwarding of the media packet;

11. A, B, and C can send the application for the right to speak to the online server, and the online server will allocate the right to speak on a first-come, first-served basis, and record the current owner of the right to speak;

12. Repeat steps 1 to 11;

13. When one party exits, send a message to the online server, and the online server notifies the broadband gateway, and the broadband gateway adjusts the media packet forwarding mechanism; when there are only two parties remaining in the conversation, one party hangs up and sends a message to the online server, and the online server notifies the other party to hang up ; At the same time, the online server sends a call end message to the broadband gateway, and the broadband gateway deletes the call information.

The system structure of the invention is simple, and the system has good practicability and expandability. SIP-MESSAGE is used between the client and the server to exchange call signaling, and the media information is described using the standard SDP protocol. Under the cooperation of PTT client, online server and broadband gateway, "one-to-one" or "one-to-many" PTT calls between users in different network domains can be realized. The invention is not only suitable for PTT calls between public network users, but also supports private network users with network address conversion function. In the present invention, media streams between public network users are directly sent to each other; for private network users, the interaction of media streams can be realized with the participation of broadband gateways, especially when performing "one-to-many" PTT calls, The calling party only needs to send the media stream to the broadband gateway, and the broadband gateway will forward the media to each called party according to the PTT call information.

Compared with the prior art, the invention inherits the traditional advantages of the PTT call, and the call establishment is simple and quick. Simultaneously, because the media packet transmission among the present invention adopts RTP/RTCP agreement to realize, greatly improved the real-time performance of media transmission, in addition, because the reliability of broadband gateway is higher, the reliability of PTT call is effectively guaranteed .

Claims (10)

1, a kind of PTT business realizing system based on voip technology comprises the PTT client, is used to initiate/accept the PTT call request, and carries out the encoding and decoding of Media Stream, it is characterized in that, also comprises:

User's line server is used to accept the registration of described client, the connection attribute of record client, and transmit the PTT call request in view of the above, in the PTT of having only the public network client to participate in is called out, assist directly to set up between the described client media channel, switched-media bag;

Broadband gateway equipment, link to each other with described user's line server by internal interface, the PTT that is used for participating in the private network client with network address translation apparatus is called out, carrying out signaling with described line server consults, and be that described client is set up media channel as intermediary, the switched-media bag.

2, the PTT business realizing system based on voip technology as claimed in claim 1 is characterized in that described public network client, comprises the public network user on the internet, and is positioned at same local area network users; Described private network client is not in same local area network users.

3, a kind of PTT service implementation method based on voip technology is characterized in that comprising the steps:

(1) the PTT client is registered to user's line server;

(2) PTT caller originating end sends call request to described line server;

(3) described line server belongs to public network user according to user's network attribute in this call request judgement call group still private user;

(4) described line server to called end forwarded call signaling, assists to set up the media channel between the user according to network attribute, the switched-media bag,

Wherein, for the calling of having only public network user to participate in,, directly set up media channel between the user, the switched-media bag by described line server forwarded call signaling;

For the calling that the private user with network address translation function participates in, by described line server forwarded call signaling, and carry out signaling with broadband gateway and consult, set up media channel via broadband gateway, the switched-media bag.

4, the PTT service implementation method based on voip technology as claimed in claim 3, it is characterized in that described step (1), when client is registered to user's line server, the connection attribute of oneself need be sent to line server, connection attribute comprises: the IP address of client place computer and the port numbers of connection.

5, the PTT service implementation method based on voip technology as claimed in claim 3, it is characterized in that described step (3), line server is to register employed address and port numbers according to the connection attribute of client and client to line server, comes the network attribute of user side in the call group is judged.

6, the PTT service implementation method based on voip technology as claimed in claim 3 is characterized in that the described calling of having only public network user to participate in, and its processing procedure comprises the steps:

(1) calling terminal with the SDP protocol description, and is encapsulated into the media information of oneself among the SIP-MESSAGE message with it, sends to line server;

(2) line server is according to the state of this message inspection called end, if called end is in busy state, then directly send call failure message to calling terminal, otherwise, line server is transmitted to called end with this call request, writes down the owner of present speaking right simultaneously;

(3) called end receives after the call request, opens media channel, prepares to receive the medium bag that comes from calling terminal, and the decoding broadcast, simultaneously call is encapsulated among the SIP-MESSAGE message, and line server is given in loopback;

(4) line server sends to calling terminal with the call message of called end;

(5) calling terminal receives after the call of called end, begins to gather local media, sends to called end after compressed encoding;

(6) after the calling terminal speech finishes, stop the transmission of medium bag, send message to line server simultaneously and discharge speaking right, line server notice called end;

(7) calling terminal and called end send the speaking right application to line server, distribute speaking right by line server according to " First come first served " principle, and write down the owner of present speaking right;

(8) repeating step (1)～(7);

(9) behind the end of conversation, on-hook side sends on-hook message to line server, notifies the opposing party on-hook by line server.

7, the PTT service implementation method based on voip technology as claimed in claim 3 is characterized in that the calling that described private user participates in, and its processing procedure comprises the steps:

(1) calling terminal is described the media information of oneself with SDP, and it is encapsulated among the SIP-MESSAGE message, then the network address translation apparatus of this message by this local area network (LAN) is sent to line server, writes down the owner of present speaking right simultaneously;

(2) line server is checked the state of called end, if called end is in busy state, then send call failure message to calling terminal, otherwise, line server is transmitted to the network address translation apparatus of called end local area network (LAN) of living in this call request, and sends to called end by this network address translation apparatus;

(3) line server sends call information to broadband gateway;

(4) broadband gateway metered call information, and prepare to receive and transmit the medium bag that comes from calling terminal;

(5) called end receives after the call request, opens media channel, prepares to receive the medium bag that comes from calling terminal, and the decoding broadcast, simultaneously call is encapsulated among the SIP-MESSAGE message, gives line server by the network address translation apparatus loopback;

(6) called end sends the UDP message to broadband gateway, so that get through broadband gateway is transmitted routing channel from medium to called end;

(7) line server sends to calling terminal with the call message of the called end network address translation apparatus by calling terminal;

(8) calling terminal receives after the call of called end, begins to gather local media, sends to broadband gateway by network address translation apparatus after compressed encoding;

(9) broadband gateway is transmitted the medium bag of calling terminal to called end;

(10) after the calling terminal speech finishes, stop the transmission of medium bag, send message to line server simultaneously and discharge speaking right, line server notice called end;

(11) calling terminal and called end send the speaking right application to line server, distribute speaking right by line server according to arriving first " getting principle earlier ", and write down the owner of present speaking right;

(12) repeating step (1)～(11);

(13) behind the end of conversation, on-hook side sends on-hook message to line server, notifies the opposing party on-hook by line server, and simultaneously, line server sends end of calling message to broadband gateway, and broadband gateway is deleted this call information.

8, as claim 6 or 7 described PTT service implementation method, it is characterized in that described called end can comprise a plurality of PTT clients based on voip technology.

9, the PTT service implementation method based on voip technology as claimed in claim 3 is characterized in that described medium bag, comprises audio stream and video flowing.

10, the PTT service implementation method based on voip technology as claimed in claim 3 is characterized in that described switched-media bag step, uses the RTP/RTCP agreement to finish.

Images