CN110971582B - Central level voice interconnection method and system between 800M and 350M systems - Google Patents

Central level voice interconnection method and system between 800M and 350M systems Download PDF

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Publication number
CN110971582B
CN110971582B CN201811162064.6A CN201811162064A CN110971582B CN 110971582 B CN110971582 B CN 110971582B CN 201811162064 A CN201811162064 A CN 201811162064A CN 110971582 B CN110971582 B CN 110971582B
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sip
communication system
call
voice stream
voice
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CN110971582A (en
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柯理理
丁瞻远
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CRSC Communication and Information Group Shanghai Co Ltd
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CRSC Communication and Information Group Shanghai Co Ltd
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    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
    • H04L65/00—Network arrangements, protocols or services for supporting real-time applications in data packet communication
    • H04L65/1066—Session management
    • H04L65/1101—Session protocols
    • H04L65/1104—Session initiation protocol [SIP]
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
    • H04L63/00—Network architectures or network communication protocols for network security
    • H04L63/08—Network architectures or network communication protocols for network security for authentication of entities
    • H04L63/0815—Network architectures or network communication protocols for network security for authentication of entities providing single-sign-on or federations
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
    • H04L65/00—Network arrangements, protocols or services for supporting real-time applications in data packet communication
    • H04L65/10—Architectures or entities
    • H04L65/102—Gateways
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
    • H04L65/00—Network arrangements, protocols or services for supporting real-time applications in data packet communication
    • H04L65/1066—Session management
    • H04L65/1083—In-session procedures
    • H04L65/1086—In-session procedures session scope modification
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
    • H04L65/00—Network arrangements, protocols or services for supporting real-time applications in data packet communication
    • H04L65/40—Support for services or applications
    • H04L65/4061—Push-to services, e.g. push-to-talk or push-to-video
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
    • H04L65/00—Network arrangements, protocols or services for supporting real-time applications in data packet communication
    • H04L65/60—Network streaming of media packets
    • H04L65/61—Network streaming of media packets for supporting one-way streaming services, e.g. Internet radio
    • H04L65/611—Network streaming of media packets for supporting one-way streaming services, e.g. Internet radio for multicast or broadcast
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
    • H04L65/00—Network arrangements, protocols or services for supporting real-time applications in data packet communication
    • H04L65/60—Network streaming of media packets
    • H04L65/65—Network streaming protocols, e.g. real-time transport protocol [RTP] or real-time control protocol [RTCP]

Abstract

The application relates to a method and a system for center-level voice interconnection and interworking between 800M and 350M systems, wherein the method comprises the following steps: step S1: reading SIP configuration and configuration information, and system information of an 800M system and a 350M system; step S2: establishing a session in SIP according to the pre-configured configuration information, and establishing communication with an 800M digital trunking communication system and communication with a 350M analog trunking communication system; step S3: if the incoming call information of the 800M digital trunking communication system or the 350M analog trunking communication system is detected, then executing step S4, and if the incoming call information of the SIP client is detected, then executing step S7; step S4: if the call type is incoming call, executing step S5, if the call is ended, executing step S6; step S5: transmitting a voice stream; step S6: and terminating the transmission of the voice stream; step S7: and receiving the opposite terminal SIP Message and the voice stream and transmitting the opposite terminal SIP Message and the voice stream to a corresponding communication system. Compared with the prior art, the method has the advantages of flexible configuration, strong expansibility and the like.

Description

Central level voice interconnection method and system between 800M and 350M systems
Technical Field
The application relates to a voice scheduling technology, in particular to a method and a system for center-level voice interconnection and interworking between 800M and 350M systems.
Background
Within some systems there are two sets of voice communication systems, for example, a 350M analog trunked communication system provided by some manufacturers and an 800M digital trunked communication system provided by european aerospace. The 350M system is established earlier, so that the whole communication of an early government network is realized, and an 800M digital trunking communication system is adopted in the subsequent network construction along with the increase of technology and the improvement of requirements. In order to realize the voice intercommunication between the 800M digital trunking communication system and the reserved 350M analog trunking communication system, so as to ensure that police officers and security personnel of a dispatch place with public security in a subway can accept the dispatching command, the interconnection and intercommunication between the two different systems in the same network must be realized.
Disclosure of Invention
The application aims to overcome the defects of the prior art and provide a method and a system for center-level voice interconnection and interworking between 800M and 350M systems.
The aim of the application can be achieved by the following technical scheme:
a center level voice interconnection method between 800M and 350M systems comprises the following steps:
step S1: reading SIP configuration and configuration information, and system information of an 800M digital trunked communication system and a 350M analog trunked communication system;
step S2: establishing a session in SIP according to the pre-configured configuration information, and establishing communication with an 800M digital trunking communication system and communication with a 350M analog trunking communication system;
step S3: simultaneously detecting whether a call exists between the 800M digital trunking communication system and the 350M analog trunking communication system and the incoming call information of the SIP client, if the incoming call information of the 800M digital trunking communication system or the 350M analog trunking communication system is detected, then executing the step S4, and if the incoming call information of the SIP client is detected, then executing the step S7;
step S4: judging the call type, if the call type is incoming call, executing step S5, and if the call type is call end, executing step S6;
step S5: initiating PTT to press the SIP Message by using the pre-bound SIP client and transmitting the voice stream;
step S6: initiating a Session Initiation Protocol (SIP) Message released by the PTT by using a pre-bound SIP client, and terminating the transmission of the voice stream;
step S7: and receiving the opposite terminal SIP Message and the voice stream and transmitting the opposite terminal SIP Message and the voice stream to a corresponding communication system.
The step S7 specifically includes:
step S71: receiving an opposite terminal SIP Message and a voice stream;
step S72: judging the type of the communication system pre-bound by the SIP client, if the communication system is an 800M digital cluster communication system, executing the step S74, and if the communication system is a 350M analog cluster communication system, executing the step S73;
step S73: transmitting the voice stream code;
step S74: and designating 800M talk groups on the designated TCS client according to the SIP Message to perform PTTup and PTT down operations, and transcoding and transmitting the fetched voice stream.
The judging basis in the step S4 specifically comprises the following steps:
for a call from an 800M digital trunking communication system, judging the type of the call according to the event mode of call information;
for a call from a 350M analog trunked communication system, the first time a voice stream is received is considered to be the originating call, and if the voice stream is not received within 100 milliseconds, the call is considered to be the ending.
The step S2 specifically includes:
step S21: establishing a session of the SIP according to the preconfigured configuration information;
step S22: establishing communication with the 800M system through the SDK provided by the 800M system, loading and using talk group information, and monitoring voice flow on a designated IP channel;
step S23: and establishing communication through a private protocol provided by the 350M system, logging in a 350M gateway by adopting the allocated seat number, and loading the talk group on the designated gateway.
The method further comprises the steps of:
and the SIP ID allocated to each physical link establishes a group call configuration relation table according to the interconnected conversation group ID and the physical link associated with the conversation group ID, and the group call SIP gateway acquires the group call configuration relation table after being started.
A center level voice interconnection and interworking system between 800M and 350M systems, comprising an 800M digital trunked communication system and a 350M analog trunked communication system, and further comprising a control server connected to the 800M digital trunked communication system and the 350M analog trunked communication system, respectively, the control server comprising a memory and a processor, and a program stored in the memory and executed by the processor, the processor implementing the following steps when executing the program:
step S1: reading SIP configuration and configuration information, and system information of an 800M digital trunked communication system and a 350M analog trunked communication system;
step S2: establishing a session in SIP according to the pre-configured configuration information, and establishing communication with an 800M digital trunking communication system and communication with a 350M analog trunking communication system;
step S3: simultaneously detecting whether a call exists between the 800M digital trunking communication system and the 350M analog trunking communication system and the incoming call information of the SIP client, if the incoming call information of the 800M digital trunking communication system or the 350M analog trunking communication system is detected, then executing the step S4, and if the incoming call information of the SIP client is detected, then executing the step S7;
step S4: judging the call type, if the call type is incoming call, executing step S5, and if the call type is call end, executing step S6;
step S5: initiating PTT to press the SIP Message by using the pre-bound SIP client and transmitting the voice stream;
step S6: initiating a Session Initiation Protocol (SIP) Message released by the PTT by using a pre-bound SIP client, and terminating the transmission of the voice stream;
step S7: and receiving the opposite terminal SIP Message and the voice stream and transmitting the opposite terminal SIP Message and the voice stream to a corresponding communication system.
The step S7 specifically includes:
step S71: receiving an opposite terminal SIP Message and a voice stream;
step S72: judging the type of the communication system pre-bound by the SIP client, if the communication system is an 800M digital cluster communication system, executing the step S74, and if the communication system is a 350M analog cluster communication system, executing the step S73;
step S73: transmitting the voice stream code;
step S74: and designating 800M talk groups on the designated TCS client according to the SIP Message to perform PTTup and PTT down operations, and transcoding and transmitting the fetched voice stream.
The judging basis in the step S4 specifically comprises the following steps:
for a call from an 800M digital trunking communication system, judging the type of the call according to the event mode of call information;
for a call from a 350M analog trunked communication system, the first time a voice stream is received is considered to be the originating call, and if the voice stream is not received within 100 milliseconds, the call is considered to be the ending.
The step S2 specifically includes:
step S21: establishing a session of the SIP according to the preconfigured configuration information;
step S22: establishing communication with the 800M system through the SDK provided by the 800M system, loading and using talk group information, and monitoring voice flow on a designated IP channel;
step S23: and establishing communication through a private protocol provided by the 350M system, logging in a 350M gateway by adopting the allocated seat number, and loading the talk group on the designated gateway.
The processor when executing the program also realizes the following steps:
and the SIP ID allocated to each physical link establishes a group call configuration relation table according to the interconnected conversation group ID and the physical link associated with the conversation group ID, and the group call SIP gateway acquires the group call configuration relation table after being started.
Compared with the prior art, the application has the following beneficial effects:
1) The scheme adopts the SIP mode, thus supporting distributed construction, flexible configuration and strong expansibility.
2) For the system side, the scheduling table is equivalent to a scheduling table for realizing two system platforms simultaneously. The voice link is converted into a local sound card device and a microphone, and can be used as a dispatching desk.
3) Finally, a centralized mode is adopted, for an 800M system, the mode realizes the simultaneous login of multiple TCS clients, for an original 800M system dispatching desk, the original dispatching desk is a device to log in one number, at present, the mode proves that a plurality of numbers can be used for simultaneous login, a plurality of TCS clients can simultaneously carry out operations such as calling, and the like, and can provide other ways for the 800M dispatching system to realize other services.
4) Providing some technical points for the digital processing (codec and transcoding) of the voice stream.
5) The interconnected nodes are flexibly selected.
Drawings
FIG. 1 is a schematic flow chart of main steps of the method of the application;
FIG. 2 is a block diagram of the system of the present application;
wherein: 1. 350M analog trunking communication system, 2, 800M digital trunking communication system, 3, control server.
Detailed Description
The application will now be described in detail with reference to the drawings and specific examples. The present embodiment is implemented on the premise of the technical scheme of the present application, and a detailed implementation manner and a specific operation process are given, but the protection scope of the present application is not limited to the following examples.
The system adopts a central level digital mode for interconnection and interworking, and firstly, the system can adopt a digital mode for communication. For the 800M system side, the system is used for providing the SDK, the signaling communication with the TCS server is established in a dispatching mode, and the voice communication with the 800M system TVG is established in an RTP protocol, so that a group calling function of one path can be established. For the 350M system side, a Wide custom private protocol is adopted to establish communication with a 350M system gateway, and voice communication is carried out through an RTP protocol to establish a path of group call. And then, the SIP session protocol is adopted, and one path of interconnection and intercommunication of the 800M-350M system can be completed by a pair of SIP clients.
The application adopts a distributed scheme, namely, one device establishes communication with one end system and one path of SIP client, but in order to save cost, the system side also supports a mode of completing multiple paths of simultaneous communication by one device, and for an 800M system, a mode of logging in multiple TCS clients on one device simultaneously can be adopted to realize the multiple paths of simultaneous group calling function. For the 350M side, the system gateway supports the multi-path simultaneous group call function. Therefore, one device can be used for establishing communication with the two-end systems and simultaneously using multiple paths of SIP clients to complete multiple paths of simultaneous interconnection and interworking. There is provided a center-level voice interconnection and interworking system between 800M and 350M systems, as shown in fig. 2, including an 800M digital trunking communication system 2 and a 350M analog trunking communication system 1, and further including a control server 3 connected to the 800M digital trunking communication system 2 and the 350M analog trunking communication system 1, respectively, as shown in fig. 1, the control method adopted by the control server includes:
firstly, a group calling configuration relation table is established for the SIP ID allocated to each physical link according to the interconnected conversation group ID and the physical link associated with the conversation group ID, and after the group calling SIP gateway is started, the group calling configuration relation table is obtained.
Step S1: reading SIP configuration and configuration information and system information of the 800M digital trunked communication system 2 and the 350M analog trunked communication system 1; for the conversation group ID which needs interconnection and intercommunication of two systems, the physical link which is related to the conversation group ID and the SIP ID which is distributed by the physical link, different group calling configuration relation tables are created for each group calling SIP gateway, after a user logs in the group calling SIP gateway, dial-up connection is automatically established between the corresponding SIP IDs through an SIP exchange server according to the obtained group calling configuration relation tables; the administrator can flexibly modify the group call configuration relation table. Specifically, the authorized user modifies the configuration file to establish a one-to-one group call configuration relationship for the talk group, the physical port number associated with the talk group ID and the SIP ID, and further needs to configure the IP address and port number of the SIP exchange server, TCS server information of the 800M system, the logged-in user name and password, and the group call number, and gateway information of the 350M system includes a seat number, a gateway IP, and the like.
Step S2: establishing a session of SIP according to the preconfigured configuration information, and establishing communication with the 800M digital trunked communication system 2 and communication with the 350M analog trunked communication system 1, specifically comprising:
step S21: establishing a session of the SIP according to the preconfigured configuration information;
step S22: establishing communication with the 800M system through the SDK provided by the 800M system, loading and using talk group information, and monitoring voice flow on a designated IP channel;
step S23: and establishing communication through a private protocol provided by the 350M system, logging in a 350M gateway by adopting the allocated seat number, and loading the talk group on the designated gateway.
Establishing communication with an 800M system, comprising the steps of simultaneously logging in by using a plurality of accounts by adopting an SDK provided by the system, establishing a communication flow with a TCS server of the 800M system, establishing a plurality of TCS clients, and selecting a group number to be used as interconnection in each TCS client to load as a main resource. Establishing communication with the 350M system includes using the internal protocol provided by the system, logging in to the 350M gateway using the system assigned agent number and establishing communication with the gateway, and then loading the required interconnected group number (also referred to as the institute number in the 350M system) on the corresponding gateway. Namely: after the interconnection server is started, firstly establishing a session of Session Initiation Protocol (SIP) according to the information of a configuration table, wherein two SIP numbers are a pair; then establishing communication with the 800M system through the SDK provided by the 800M system, loading and using talk group information, and monitoring voice flow on a designated IP channel; and finally, establishing communication through a private protocol provided by the 350M system, logging in a 350M gateway by adopting the allocated seat number, and loading the talk group on the designated gateway.
Step S3: meanwhile, detecting whether a call exists in the 800M digital trunking communication system 2 and the 350M analog trunking communication system 1 and the incoming call information of the SIP client, if the incoming call information of the 800M digital trunking communication system 2 or the 350M analog trunking communication system 1 is detected, then executing the step S4, and if the incoming call information of the SIP client is detected, then executing the step S7;
the device detects that an incoming call signaling exists at a certain TCS client of the 800M system and simultaneously takes a corresponding voice stream through an RTP protocol, because no additional information is needed for 350M incoming call (such as PTT is pressed down, etc.), the voice stream is directly transcoded and directly sent to an opposite-end SIP client through the corresponding SIP client, the opposite-end SIP client transcodes after obtaining voice (converts into a code stream needed by the 350M system), and then sends to a 350M voice gateway through the RTP protocol of the end, thereby realizing the function of answering the 350M system outgoing from the 800M system.
Step S4: judging the call type, if the call type is an incoming call, executing step S5, and if the call is ended, executing step S6, wherein the judging basis specifically comprises the following steps:
for a call from the 800M digital trunking communication system 2, judging the type of the call according to the event mode of the call information;
for a call from the 350M analog trunked communication system 1, the first time a voice stream is received is considered to be the originating call, and if the voice stream is not received within 100 milliseconds, the call is considered to be the ending.
In the step, whether the incoming call information or the call ending information is detected, and the 800M system can detect the call information of which TCS client according to the event mode of the SDK; for the 350M system, the detection is based on the voice stream, the first time the voice stream is received is regarded as the call originating from the opposite end, and if the voice stream is not received within 100 ms, the call of the opposite end is regarded as the call ending.
Step S5: initiating PTT to press the SIP Message by using the pre-bound SIP client and transmitting the voice stream;
step S6: initiating a Session Initiation Protocol (SIP) Message released by the PTT by using a pre-bound SIP client, and terminating the transmission of the voice stream;
the device firstly sends an SIP message representing the PTT of the opposite terminal to the SIP client of the opposite terminal through the SIP client when detecting that a certain group of the 350M system has an incoming call (by detecting that the corresponding RTP protocol has voice data), and the SIP client presses the PTT of the loaded group on the corresponding TCS client (communicates with the TCS server of the 800M system) after receiving the SIP message; and then the SIP client at the 350M system side transcodes and transmits the obtained voice stream, and the opposite-end SIP client transcodes the received voice stream and transmits the voice stream to the TVG server through RTP. Thus, the function of answering the 350M outgoing 800M system is realized. When no voice is detected for more than 100 ms, a SIP message of the counterpart PTT release is sent through the SIP client. The opposite SIP client detects the message and releases PTT on the corresponding TCS client.
Step S7: receiving opposite-end SIP Message and voice stream, and transmitting to a corresponding communication system, specifically comprising:
step S71: receiving an opposite terminal SIP Message and a voice stream;
step S72: judging the type of the communication system pre-bound by the SIP client, if the communication system is the 800M digital trunking communication system 2, executing the step S74, and if the communication system is the 350M analog trunking communication system 1, executing the step S73;
step S73: transmitting the voice stream code;
step S74: PTTup and PTT down operations are performed (through SDK and the fetched voice stream is transcoded and sent) according to SIP Message at the designated TCS client by designating 800M talk group.
The system has a dynamic selection function, namely the SIP client provides a channel, and the 800M system and the 350M system are respectively bound to the corresponding clients, so that the 800M system can be selected to be connected with the groups when no call exists, and the 350M system can also be used for selecting the groups to be connected with the groups. Meanwhile, the SIP client can also be manually operated to select the dialed opposite-end SIP number, so that the inter-group communication of the same system can be realized.
In addition, the application is fully disclosed, the application can be realized by pure software improvement, and can be realized by software and hardware improvement, for example, by configuring a voice board card, and the model of the voice board card can adopt east K161A-E.

Claims (10)

1. A method for center-level voice interconnection between 800M and 350M systems, comprising:
step S1: reading SIP configuration and configuration information, and system information of an 800M digital trunked communication system (2) and a 350M analog trunked communication system (1), comprising: TCS server information of 800M system, logged user name and password and group call number, gateway information of 350M system includes seat number, gateway IP;
step S2: establishing a SIP session according to the preconfigured configuration information, and establishing communication with the 800M digital trunking communication system (2) and communication with the 350M analog trunking communication system (1);
establishing communication with an 800M system, comprising: the SDK provided by the system is used for logging in simultaneously by using a plurality of accounts, a communication flow with a TCS server of the 800M system is established in a scheduling mode, voice communication with a TVG of the 800M system is established through an RTP protocol, a plurality of TCS clients are established, and a group number to be used as interconnection is selected from all TCS clients to be loaded as a main resource;
establishing communication with a 350M system, including adopting an internal protocol provided by the 350M system, logging in the communication of a 350M gateway by using the seat number distributed by the system, establishing the communication with the gateway, and loading the required interconnected group number on the corresponding gateway;
adopting a Session Initiation Protocol (SIP), and completing interconnection and interworking of one path of an 800M-350M system by using a pair of SIP clients;
step S3: simultaneously detecting whether a call exists in the 800M digital trunking communication system (2) and the 350M analog trunking communication system (1) and the incoming call information of the SIP client, if the incoming call information of the 800M digital trunking communication system (2) or the 350M analog trunking communication system (1) is detected, then executing the step S4, and if the incoming call information of the SIP client is detected, executing the step S7;
when detecting that a certain TCS client of the 800M system has incoming signaling and simultaneously takes a corresponding voice stream through an RTP protocol, transcoding the voice stream and directly sending the voice stream to an opposite-end SIP client through the corresponding SIP client, transcoding the opposite-end SIP client after obtaining voice, and sending the voice stream to a 350M voice gateway through the RTP protocol of the end;
when detecting that a certain group of the 350M system has an incoming call, sending an SIP message representing the PTT press of the opposite end to the opposite end SIP client through the SIP client, and after the SIP client receives the SIP message, pressing the loaded group on the corresponding TCS client to communicate with a TCS server of the 800M system; then the SIP client at 350M system side transcodes and transmits the obtained voice stream, the opposite end SIP client transcodes the received voice stream and transmits the voice stream to the TVG server through RTP;
step S4: judging the call type, if the call type is incoming call, executing step S5, and if the call type is call end, executing step S6;
step S5: initiating PTT to press the SIP Message by using the pre-bound SIP client and transmitting the voice stream;
step S6: initiating a Session Initiation Protocol (SIP) Message released by the PTT by using a pre-bound SIP client, and terminating the transmission of the voice stream;
step S7: and receiving the opposite terminal SIP Message and the voice stream and transmitting the opposite terminal SIP Message and the voice stream to a corresponding communication system.
2. The method for interworking between 800M and 350M systems according to claim 1, wherein the step S7 specifically includes:
step S71: receiving an opposite terminal SIP Message and a voice stream;
step S72: judging the type of the communication system pre-bound by the SIP client, if the communication system is an 800M digital trunked communication system (2), executing step S74, and if the communication system is a 350M analog trunked communication system (1), executing step S73;
step S73: transmitting the voice stream code;
step S74: and designating 800M talk groups on the designated TCS client according to the SIP Message to perform PTT up and PTT down operations, and transcoding and transmitting the fetched voice stream.
3. The method for interworking between 800M and 350M systems according to claim 1, wherein the determining in step S4 specifically comprises:
for a call from an 800M digital trunking communication system (2), judging the type of the call according to the event mode of call information;
for a call from the 350M analog trunked communication system (1), the first time a voice stream is received is considered to originate the call, and if the voice stream is not received within 100 milliseconds, the call is considered to be ended.
4. The method for interworking between 800M and 350M systems according to claim 1, wherein the step S2 specifically includes:
step S21: establishing a session of the SIP according to the preconfigured configuration information;
step S22: establishing communication with the 800M system through the SDK provided by the 800M system, loading and using talk group information, and monitoring voice flow on a designated IP channel;
step S23: and establishing communication through a private protocol provided by the 350M system, logging in a 350M gateway by adopting the allocated seat number, and loading the talk group on the designated gateway.
5. The method for center-level voice interconnection between 800M and 350M systems according to claim 1, further comprising:
and the SIP ID allocated to each physical link establishes a group call configuration relation table according to the interconnected conversation group ID and the physical link associated with the conversation group ID, and the group call SIP gateway acquires the group call configuration relation table after being started.
6. A center-level voice interconnection and interworking system between 800M and 350M systems, comprising an 800M digital trunked communication system (2) and a 350M analog trunked communication system (1), characterized by further comprising a control server (3) connected to the 800M digital trunked communication system (2) and the 350M analog trunked communication system (1), respectively, the control server (3) comprising a memory and a processor, and a program stored in the memory and executed by the processor, the processor implementing the steps when executing the program of:
step S1: reading SIP configuration and configuration information, and system information of an 800M digital trunked communication system (2) and a 350M analog trunked communication system (1);
step S2: establishing a SIP session according to the preconfigured configuration information, and establishing communication with the 800M digital trunking communication system (2) and communication with the 350M analog trunking communication system (1);
step S3: simultaneously detecting whether a call exists in the 800M digital trunking communication system (2) and the 350M analog trunking communication system (1) and the incoming call information of the SIP client, if the incoming call information of the 800M digital trunking communication system (2) or the 350M analog trunking communication system (1) is detected, then executing the step S4, and if the incoming call information of the SIP client is detected, executing the step S7;
step S4: judging the call type, if the call type is incoming call, executing step S5, and if the call type is call end, executing step S6;
step S5: initiating PTT to press the SIP Message by using the pre-bound SIP client and transmitting the voice stream;
step S6: initiating a Session Initiation Protocol (SIP) Message released by the PTT by using a pre-bound SIP client, and terminating the transmission of the voice stream;
step S7: and receiving the opposite terminal SIP Message and the voice stream and transmitting the opposite terminal SIP Message and the voice stream to a corresponding communication system.
7. The system for central voice interconnection and interworking between 800M and 350M systems according to claim 6, wherein the step S7 specifically comprises:
step S71: receiving an opposite terminal SIP Message and a voice stream;
step S72: judging the type of the communication system pre-bound by the SIP client, if the communication system is an 800M digital trunked communication system (2), executing step S74, and if the communication system is a 350M analog trunked communication system (1), executing step S73;
step S73: transmitting the voice stream code;
step S74: and designating 800M talk groups on the designated TCS client according to the SIP Message to perform PTT up and PTT down operations, and transcoding and transmitting the fetched voice stream.
8. The system for interworking between 800M and 350M systems according to claim 6, wherein the determining in step S4 specifically comprises:
for a call from an 800M digital trunking communication system (2), judging the type of the call according to the event mode of call information;
for a call from the 350M analog trunked communication system (1), the first time a voice stream is received is considered to originate the call, and if the voice stream is not received within 100 milliseconds, the call is considered to be ended.
9. The system for interworking between 800M and 350M systems according to claim 6, wherein the step S2 specifically comprises:
step S21: establishing a session of the SIP according to the preconfigured configuration information;
step S22: establishing communication with the 800M system through the SDK provided by the 800M system, loading and using talk group information, and monitoring voice flow on a designated IP channel;
step S23: and establishing communication through a private protocol provided by the 350M system, logging in a 350M gateway by adopting the allocated seat number, and loading the talk group on the designated gateway.
10. The system of claim 6, wherein the processor, when executing the program, further performs the steps of:
and the SIP ID allocated to each physical link establishes a group call configuration relation table according to the interconnected conversation group ID and the physical link associated with the conversation group ID, and the group call SIP gateway acquires the group call configuration relation table after being started.
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CN1585520A (en) * 2004-06-04 2005-02-23 中兴通讯股份有限公司 Interconnecting and interflowing method for digital cluster system and common telephone system
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