TWI876197B - High-availability multimedia gateway system, multimedia gateway management method and computer-readable medium - Google Patents

Abstract

A high-availability multimedia gateway system, a multimedia gateway management method and a computer-readable medium are provided, which dynamically activate or deactivate multimedia gateways according to the resource usage of the multimedia gateways to adjust system resources. In addition, if a first multimedia gateway fails or its resources are insufficient, a second multimedia gateway is controlled to seamlessly take over the ongoing sessions of the first multimedia gateway in the present invention, so as to ensure the quality of multimedia services and communication services.

Description

High availability multimedia gateway system, multimedia gateway management method and computer readable media

The present invention relates to a high-availability multimedia gateway system, a multimedia gateway management method and a computer-readable medium, which utilizes a call automatic allocation mechanism to handle instantaneous large-scale calls for accessing the same voice content, providing high-availability call quality, and through highly efficient call conference packet analysis and storage technology, as well as call conference reconstruction and takeover technology, different multimedia gateways can coordinate the configuration of system resources, take over or replace the original multimedia gateway's call conference, and maintain the call conference without interruption and provide stable and accurate voice content, so as to ensure the quality and best experience of users dialing and accessing real-time voice content.

Currently, the common multimedia gateways are physical devices. Due to the limitation of hardware, they cannot be flexibly expanded according to the call volume. When the call volume is not large, it will waste system resources. When the call volume increases instantly, there is a fear that the system resources cannot cope with it, which may cause the client to be unable to obtain real-time voice content. In addition, when the equipment fails, the call on the client side will be interrupted due to the switching of lines. If the interruption occurs during an important video conference, the call of the participants will be interrupted, causing inconvenience. In order to take into account the optimization of system resource utilization and the demand for high-availability real-time call conferences, the technology of providing high-availability multimedia gateways that automatically adjust according to traffic has begun to be proposed.

However, the current various commonly used technologies cannot dynamically take over the call information of other SIP servers at the back-end Session Initiation Protocol (SIP) server, and cannot arbitrarily transfer the data of this call to other SIP servers. Therefore, when the traffic is large, load balancing can only be performed through the pre-configured SIP server; conversely, when the traffic is small, the resources of the pre-configured SIP server cannot be transferred to other uses. This is not a good design and needs to be improved urgently.

The purpose of this invention is to provide a high-availability multimedia gateway system and multimedia gateway management method to address the deficiencies of the above-mentioned conventional technologies. The system does not require the configuration of a large number of physical devices, but uses cloud computing and hardware virtualization software technology to configure multimedia gateways.

The present invention activates the pre-configured multimedia gateway virtual machine according to the call flow threshold, dynamically adds or shuts down the multimedia gateways that provide online services, or generates simulated packets of the initial call conversation for the multimedia gateway to rebuild the conversation on other multimedia gateways for taking over. The call conversation can also be seamlessly transferred or concentrated to the same multimedia gateway to reduce the consumption of system resources.

The system for achieving the above-mentioned invention purpose includes a multimedia signal generation system and a high-availability multimedia gateway system.

The multimedia signal generation system can provide time signal audio source from the National Time and Frequency Standards Laboratory, or real-time voice broadcast audio source from other equipment vendors or operators. The analog audio source can be directly connected to the multimedia gateway through the Registered Jack-9 (RJ9), Registered Jack-11 (RJ11), Registered Jack-45 (RJ45) or Bayonet Neill-Concelman (BNC) connector; in terms of video images, the multimedia image provider can provide real-time images and connect to the multimedia gateway through the network interface. The real-time image can be a government policy propaganda video or an advertising video provided by an enterprise.

The above-mentioned high-availability multimedia gateway system can accept analog audio sources generated by multimedia systems, provide driver components for sound cards through Advanced Linux Sound Architecture (ALSA) to process the sound devices in the system, and can also accept images generated by multimedia systems. It uses the signal exchange core SIP/Real-time Transport Protocol (RTP) to receive calls from SIP application servers externally, and integrates the following functional modules internally:

High availability distribution module, used to receive the session packets from the external SIP application server and distribute the traffic flow of the call session packets.

The traffic analysis and processing module is used to store the packet field information required for call conference reestablishment and receive resource usage statistics of all multimedia gateways under the management and monitoring module to notify the high availability allocation module of the information.

The management and monitoring module is used to manage the system resources of all multimedia gateways and dynamically adjust the number of multimedia gateway machines.

The signal reconstructing module is used to provide the packets required to reconstruct the conversation.

Database used to record packet information during a call conversation and provide packet information fields required to reconstruct the conversation.

The multimedia gateway is used to convert analog audio sources into digital audio sources through a Peripheral Component Interconnect Express (PCI-E) sound card. It can also receive images from a video capture card and network audio and video images, and has SIP/RTP protocol and signal recovery functions.

The multimedia gateway management method for achieving the above-mentioned invention purpose includes a call conference recording and takeover setting process, a conference reconstruction conflict setting process, a non-sequential conference reconstruction setting process, and a multimedia gateway automatic adjustment high availability continuous call setting process.

Call conference recording and takeover setting process: The client starts the call, records the traffic packets during the call conference, and analyzes and processes them. When the multimedia gateway fails, the second multimedia gateway is activated to take over the call conference of the first multimedia gateway. The client can maintain the original call status, that is, complete the takeover setting without interrupting the call.

Conference reestablishment conflict setting process: Because the network environment of multimedia gateway configurations is different, during the conference reestablishment process, individual conference calls may use the same resources and cause conflicts. This process includes the setting method for correcting resource conflicts.

Non-sequential conference re-establishment setup process: It does not need to follow the original call conference sequence, and the call conference can also be re-established in reverse through the packet reassembly mechanism.

Multimedia gateway automatically adjusts the high availability and uninterrupted call setting process: Based on the traffic flow of the current call and conference packets, the multimedia gateway can seamlessly take over the call and conference of the original multimedia gateway when configuring system resources, so that the client can maintain the call and conference without interruption, and provide stable voice and accurate image content to ensure that users can dial and access real-time multimedia content.

On the other hand, the present invention provides a high-availability multimedia gateway system, comprising: a plurality of multimedia gateways for providing a session initiation protocol (SIP) call conversation function; a high-availability allocation module for allocating the call conversation to a first multimedia gateway among the multimedia gateways when a client initiates a call conversation; a traffic analysis and processing module for extracting establishment information of the call conversation from a traffic packet of the call conversation to store the establishment information; a management monitoring module for monitoring the activity of the first multimedia gateway during the call session to detect whether there is any abnormality, so as to make the second multimedia gateway among the multimedia gateways take over the call session when there is any abnormality in the activity of the first multimedia gateway; and a service reconstruction module for generating a plurality of session reconstruction packets of the call session according to the establishment information, so as to allow the second multimedia gateway to reconstruct the state of the call session when taking over the call session.

In one embodiment, the management monitoring module is reused to monitor and provide resource usage of the multimedia gateways, and the high availability allocation module allocates the call conference to the first multimedia gateway based on the resource usage.

In one embodiment, if the resources of the first multimedia gateway are insufficient to handle the call conversation, or the first multimedia gateway fails, the management monitoring module determines that the activity of the first multimedia gateway is abnormal.

In one embodiment, the management monitoring module is reused to activate or deactivate at least one of the multimedia gateways according to the resource usage of the multimedia gateways to adjust the resource amount of the multimedia gateways.

In one embodiment, the establishment information of the call session includes the SIP INVITE packets of the ingress and egress of the calling end and the called end of the call session, the SIP fields and the Session Description Protocol (SDP) fields of the SIP 200 OK packet and the SIP ACK packet, and the paired relationship between the ingress and egress of the calling end and the called end.

In one embodiment, the establishment information of the call session includes packets stored in mutually converted forms between SIP INVITE packets and SIP 200 OK packets, so that the second multimedia gateway does not need to follow the original packet sequence of the call session when reconstructing the state of the call session based on the packets stored in mutually converted forms.

In one embodiment, the second multimedia gateway simulates the sending and receiving of the session reestablishment packets to reestablish the state of the call session.

In one embodiment, the traffic reorganization module is reused to check whether the communication port allocated by the first multimedia gateway to the call conference has been occupied by the second multimedia gateway when the second multimedia gateway takes over the call conference. If the communication port has been occupied by the second multimedia gateway, the traffic reorganization module allocates a new communication port that is not occupied by the second multimedia gateway to the call conference after it is taken over.

In another aspect, the present invention provides a multimedia gateway management method, comprising: when a user terminal initiates a call session of a conference initiation protocol, allocating the call session to a first multimedia gateway among a plurality of multimedia gateways; extracting establishment information of the call session from a traffic packet of the call session to store the establishment information; during the call session, monitoring the first multimedia gateway; Whether the activity of a multimedia gateway is abnormal, wherein when the activity of the first multimedia gateway is abnormal, the second multimedia gateway among the multimedia gateways is allowed to take over the call conversation; and according to the establishment information, a plurality of conversation reconstruction packets of the call conversation are generated so that the second multimedia gateway can reconstruct the state of the call conversation when taking over the call conversation.

On another aspect, the present invention provides a computer-readable medium for use in a multimedia gateway system, which stores instructions for executing the above-mentioned multimedia gateway management method.

110: Multimedia signal generation system

120:SIP application server

130:SIP phone

140: Digital audio receiver

150:POTS phone

160: Mobile phone

170: Analog audio receiver

200: High availability multimedia gateway system

210~212: Multimedia gateway

220~222: Media User Agent

230~232: Service recovery processor

240: Management monitoring module

250: Communication service reorganization module

260: Database

270: High availability distribution module

280: Information analysis and processing module

301~319,410~450,510~560,610~650,710~780: Process steps

Figure 1 is a schematic diagram of the interface between the multimedia gateway and the external terminal of the high-availability multimedia gateway system and multimedia gateway management method of the present invention.

Figure 2 is a diagram of the architecture of the high-availability multimedia gateway system of the present invention.

Figure 3 is a packet traffic flow chart of the call conference takeover of the present invention.

Figure 4 is a flowchart of call conference recording and takeover settings of the multimedia gateway management method of the present invention.

Figure 5 is a flow chart of the conference reconstruction conflict setting of the multimedia gateway management method of the present invention.

Figure 6 is a flowchart of the non-sequential negotiation reconstruction setting of the multimedia gateway management method of the present invention.

Figure 7 is a flowchart of the multimedia gateway automatic adjustment high availability continuous setting of the multimedia gateway management method of the present invention.

Please refer to Figure 1, which is a schematic diagram of the interface of the high-availability multimedia gateway that automatically adjusts according to traffic of the present invention. The high-availability multimedia gateway system and multimedia gateway management method of the present invention include a multimedia signal generation system 110 and a high-availability multimedia gateway system 200 for accessing real-time voice and image through user calls. The client can use call dialing to access high-availability real-time multimedia content through protocols such as SIP/RTP.

In one embodiment, the multimedia signal generation system 110 is connected to the high-availability multimedia gateway system 200 through an interface such as RJ11, RJ45 or BNC to provide multimedia content, so that a client such as a plain old telephone service (POTS) phone 150 can access high-availability real-time multimedia content through a SIP application server 120 and the high-availability multimedia gateway system 200 using a call dialing method through protocols such as SIP/RTP.

The SIP application server 120 is responsible for forwarding SIP signals to provide advanced services, such as three-way calling services. The client of the high-availability multimedia gateway system 200 may include a SIP phone 130, a digital audio source receiver 140, a POTS phone 150, a mobile phone 160, and an analog audio source receiver 170. The analog audio source receiver 170 and the digital audio source receiver 140 can receive and process multimedia audio sources, and also include time synchronization services. Other general terminals can also access multimedia content through calls.

Please refer to FIG. 2, which is a schematic diagram of the high-availability multimedia gateway system 200 of the present invention. The high-availability multimedia gateway system 200 includes multimedia gateways 210-212, a management and monitoring module 240, a traffic reorganization module 250, a database 260, a high-availability allocation module 270, and a traffic analysis and processing module 280. It should be noted that the multimedia gateways 210-212 in FIG. 2 are only for demonstration purposes. In another embodiment, the high-availability multimedia gateway system 200 may include any number of multiple multimedia gateways.

Each multimedia gateway 210-212 includes a media user agent 220-222 and a traffic recovery processor 230-232. The multimedia gateway 210-212 is communicatively connected to the multimedia signal generation system 110, the management and monitoring module 240, the traffic reorganization module 250 and the high availability allocation module 270. The management and monitoring module 240 is communicatively connected to the multimedia gateway 210-212 and the traffic analysis processing module 280. The traffic reorganization module 250 is communicatively connected to the traffic recovery processor 230-232 and the database 260 of the multimedia gateway 210-212. The database 260 is communicatively connected to the traffic reorganization module 250 and the traffic analysis processing module 280. The high availability distribution module 270 is connected to the multimedia gateways 210-212, the traffic analysis processing module 280 and the SIP application server 120. The traffic analysis processing module 280 is connected to the management and monitoring module 240, the database 260 and the high availability distribution module 270.

In one embodiment, the multimedia gateways 210-212, the management and monitoring module 240, the traffic reorganization module 250, the database 260, the high availability allocation module 270, and the traffic analysis and processing module 280 can all be a single physical device, or a virtual machine of virtualization technologies such as VMware, KVM, Xen, and Microsoft Hyper-V, or a container of containerization technologies such as Docker.

The multimedia gateways 210-212 are used to convert the analog signals generated by the multimedia signal generation system 110 into RTP protocol digital signals to facilitate multimedia signal transmission on the Internet. The multimedia gateways 210-212 have a small SIP service function and include media user agent terminals 220-221 as SIP server terminals to provide multimedia content access services. In addition, the multimedia gateways 210-212 can be used as SIP servers to provide call functions between client terminals. The multimedia gateways 210-212 also include service recovery processors 230-232.

When multimedia gateways 210~212 are abnormal, shut down, or take over services, in addition to being responsible for rebuilding SIP/RTP signals, they must also send updated information about the changed multimedia gateway to notify the client to avoid call interruption. In addition, when both multimedia gateways are started normally, one of the multimedia gateways can take over the call conference of the other multimedia gateway. For example, when both multimedia gateways 210 and multimedia gateway 211 are started normally, multimedia gateway 211 can take over the call conference of multimedia gateway 210.

In one embodiment, a call conversation occurs between a calling terminal and a called terminal, wherein one of the calling terminal and the called terminal may be a media user agent terminal of a multimedia gateway, and the other of the calling terminal and the called terminal may be a client terminal. Alternatively, the calling terminal and the called terminal may be different client terminals.

The management and monitoring module 240 is used to manage the activation and deactivation of each multimedia gateway 210-212. The management and monitoring module 240 can receive the traffic analysis of the traffic analysis processing module 280. If the traffic is at peak usage, the management and monitoring module 240 will automatically activate more multimedia gateways 210-212 for load balancing; if the traffic is at off-peak usage, the management and monitoring module 240 will notify some multimedia gateways (such as multimedia gateway 2 11~212) to shut down, and notify the multimedia gateway to be taken over (for example, multimedia gateway 210) to activate its service recovery processor 230 to take over the call conversation of the multimedia gateway 211~212 that is about to be shut down, so that the client's call can continue without being interrupted due to the shutdown of the multimedia gateway 211~212.

In addition, the management monitoring module 240 monitors the resource usage of each multimedia gateway 210-212, and informs the high availability allocation module 270 of the resource usage through the traffic analysis processing module 280. The above resources include the computing time of the central processing unit (CPU) of the multimedia gateway, the storage space of the memory, and the network I/O resources (such as communication ports).

After receiving the request from the service recovery processor 230, the service reorganization module 250 can replace the Internet Protocol (IP) address and communication port of the taken-over multimedia gateways 211~212 in the packet exchange content of the original call session stored in the database 260 with the IP address and communication port of the taking-over multimedia gateway 210. If the communication port of the taken-over multimedia gateway conflicts with the communication port of the taking-over multimedia gateway 210, the communication port of the SDP will be changed and the client will be notified through the SIP reINVITE packet or update packet to achieve the takeover function.

The database 260 stores two data fields for call conferences, namely, the call-related fields remote tag and local tag and the packet information of the associated fields. In detail, the packets sent by the calling end and the called end of each call conference will pass through a multimedia gateway in the high-availability multimedia gateway system 200. From the perspective of the multimedia gateway, the calling end and the called end each have two conference pins, namely, ingress and egress, which correspond to the remote tag and local tag mentioned above.

Please refer to steps 301, 302 and 304-307 in Figure 3. When starting a call session, the calling end will first send an invitation (INVITE) packet to the called end, and the called end will reply with an acceptance (200 OK) packet to the calling end, and then the calling end will send an acknowledgment (ACK) packet to the called end. These packets will be forwarded by the central multimedia gateway. Therefore, the input of the calling end includes the above-mentioned INVITE packet and ACK packet, the output of the calling end includes the above-mentioned 200 OK packet, the input of the called end includes the above-mentioned 200 OK packet, and the output of the called end includes the above-mentioned INVITE packet and ACK packet.

The database 260 stores the SIP fields defined by the Request for Comment No. 3261 (RFC 3261) and the SDP fields defined by the Request for Comment No. 4566 (RFC 4566) in the incoming and outgoing packets of the calling end and the called end. The SIP fields include fields such as Via, From, To, Contact, Call-ID, and CSeq. The SDP fields include fields such as the protocol version, the session name, the RTP transport name and the transport IP location information, the RTP/Real-time Transport Control Protocol (RTCP) communication port, the call attributes, and the two-way call encoding method. In addition, the database 260 also stores the paired relationship between the input and output pins of the calling end and the called end of the same call conversation. The SIP field and SDP field of each packet of the input and output pins of the calling end and the called end, as well as the paired relationship between the conversation pins of the calling end and the called end, can be collectively referred to as the establishment information of the call conversation.

The high availability distribution module 270 receives the resource usage of each multimedia gateway from the management and monitoring module 240 through the traffic analysis processing module 280. When the high availability distribution module 270 receives a packet from the client through the SIP application server 120, it will decide to transmit the packet to the multimedia gateway with less or the least resource usage among the multimedia gateways 210-212 according to the resource usage to provide a load balancing function. In addition, call session packets between the client and the media user agent 220-222 of the multimedia gateway 210-212, as well as call session packets between the clients, will pass through the high availability distribution module 270. The high availability distribution module 270 will copy the invitation (INVITE), reINVITE, update (UPDATE), confirmation (ACK), acceptance (200 OK), options (OPTIONS) and information (INFO) packets defined in SIP RFC 3261 in these call session packets and transmit them to the traffic analysis processing module 280.

The traffic analysis processing module 280 is used to receive packet information from the high availability distribution module 270, and to store the packets into the database 260 after simplified processing according to the remote label and the local label, that is, to store the establishment information of each call session into the database 260 for the multimedia gateway 210 to perform the call recovery switching mechanism of the traffic recovery processor 230. On the other hand, because the traffic analysis processing module 280 records the load of individual multimedia gateways 210-212, when the load demand changes, it will notify the management monitoring module 240 to activate more multimedia gateways to increase system resources, or shut down some multimedia gateways to reduce system resources, so that system resources can be adjusted automatically and flexibly.

The following is a detailed description of the various method flows of the present invention, wherein the packets such as INVITE, reINVITE, UPDATE, Trying, Ringing, ACK and 200 OK, as well as other packets appearing below, are all packets defined in RFC 3261 of SIP.

Please refer to Figure 3, which is a packet traffic flow chart of call conference takeover, in which multimedia gateway 211 takes over the call conference of multimedia gateway 210 as an example. The media user agent of multimedia gateway 210 and the media user agent of multimedia gateway 211 are both SIP service terminals that can provide access services for the same multimedia content through multimedia signal generation system 110. The following describes the steps of the process of Figure 3, in which the "/SDP" suffix of SIP call packets such as INVITE/SDP and 200 OK/SDP emphasizes that these packets all contain the aforementioned SDP field.

In step 301, the client sends an INVITE/SDP packet to the multimedia gateway 210. Before the multimedia gateway 210 forwards the INVITE/SDP packet to its media user agent in step 302, it sends a 100 Trying packet in step 303 to inform the client that the INVITE/SDP packet is currently being forwarded. Because the automatic answering method is adopted, no 180 Ringing packet is generated here.

In step 304, the media user agent of the multimedia gateway 210 directly returns a 200 OK/SDP packet. In step 305, the multimedia gateway 210 forwards the 200 OK/SDP packet to the client. In step 306, the client returns an ACK packet to the multimedia gateway 210. In step 307, the multimedia gateway 210 forwards the ACK packet to its media user agent, and the call relationship has been established.

At this time, the multimedia gateway 211 can take over the call and rebuild the call session when the multimedia gateway 210 is turned off or on. Since the packet information exchanged by both parties is stored in the database 260 after processing under the multimedia gateway 210, the multimedia gateway 211 can use its traffic recovery processor 231 to reconstruct the packet information in the database 260. Then, the multimedia gateway 211 is ready to take over, and will restore the call of the multimedia gateway 210 in steps 308, 311 and 312 by simulating the packet status, so as to avoid the client receiving the restored signal and affecting the subsequent client behavior. The subsequent process is described in detail below.

First, in step 308, the multimedia gateway 211 simulates receiving the INVITE/SDP packet from the client (its content is the INVITE/SDP packet in the synchronization process 301). In step 309, the multimedia gateway 211 forwards the INVITE/SDP packet (its content is the INVITE/SDP packet in the synchronization process 302) to its media user agent. Then in step 310, the multimedia gateway 211 receives the 200 OK/SDP packet sent back by its media user agent. In step 311, the multimedia gateway 211 simulates forwarding the 200 OK/SDP packet (its content is the 200 OK/SDP packet in the synchronization process 305) to the client. In step 312, the multimedia gateway 211 simulates receiving the ACK packet sent back by the client (its content is the ACK packet in the sync 306). In step 313, the multimedia gateway 211 forwards the ACK packet (its content is the ACK packet in the sync 307) to its media user agent. At this time, both the multimedia gateway 210 and the multimedia gateway 211 have the information of the call session.

The simulated forwarding and simulated reception of steps 308, 311 and 312 are only simulations within the multimedia gateway 211. In fact, there is no real packet transmission between the multimedia gateway 211 and the client. This is to enable the multimedia gateway 211 to reconstruct the initial state of the call conversation in a way that does not affect the client. Because the client is currently talking through the multimedia gateway 210, it will not send INVITE/SDP packets and ACK packets, nor will it expect to receive 200 OK/SDP packets. Therefore, the multimedia gateway 211 can only simulate the forwarding and reception of packets. Otherwise, in step 311, if the multimedia gateway 211 actually forwards the 200 OK/SDP packet of step 310 to the client, the client will send an error message, and may even interrupt the ongoing call.

Next, to complete the takeover procedure, the multimedia gateway 211 sends reINVITE/SDP (or UPDATE/SDP) packets in steps 314 and 315 to notify the client and the media user agent of the multimedia gateway 211. Then, the multimedia gateway 211 receives and processes the 200 OK/SDP packets sent back by the client and the media user agent of the multimedia gateway 211 in steps 316 and 317 in accordance with the SIP RFC 3261 specification, and responds to the client and the media user agent of the multimedia gateway 211 with ACK packets in steps 318 and 319.

In addition, the call conference takeover process jointly executed by the management monitoring module 240, the traffic reorganization module 250, the multimedia gateway, and the traffic recovery processor of the multimedia gateway is described here.

First, the management and monitoring module 240 will monitor the resources of the multimedia gateways under its jurisdiction at any time to see if there is excess, and notify the multimedia gateways with excess resources to shut down. If the multimedia gateway with excess resources currently has call conversations in progress, these ongoing call conversations will cause the user's call to be interrupted if they are suddenly closed. Therefore, the management and monitoring module 240 notifies another multimedia gateway (hereinafter referred to as the takeover multimedia gateway) to start its traffic recovery processor. The traffic reorganization module 250 obtains the call conversation establishment information of the multimedia gateway with excess resources (hereinafter referred to as the taken-over multimedia gateway) from the database 260. The traffic reorganization module 250 will reassemble the call conference establishment information into three conference reconstruction packets, namely, INVITE, 200 OK and ACK, which are respectively for the inbound and outbound legs (such conference reconstruction packets are used in steps 308, 311 and 312 in FIG3). Since the IP addresses of the takeover multimedia gateway and the taken over multimedia gateway are different, when reorganizing the conference reconstruction packets, the traffic reorganization module 250 will update the IP address of the taken over multimedia gateway in these packets to the IP address of the takeover multimedia gateway. Then, the traffic recovery processor of the takeover multimedia gateway uses the aforementioned conference reconstruction packets to reconstruct the same call conference establishment information as that of the taken over multimedia gateway in the takeover multimedia gateway in a simulated manner. At this time, the client does not know that it needs to communicate with the receiver multimedia gateway, because the multimedia stream is still on the receiver multimedia gateway side. Therefore, the receiver multimedia gateway will send a real reINVITE/SDP or UPDATE/SDP packet to inform the client, asking the client to conduct multimedia streaming with the receiver multimedia gateway. Here, the SDP field of the reINVITE or UPDATE packet must include the IP address of the receiver multimedia gateway so that the client can switch the multimedia stream to the receiver multimedia gateway.

Please refer to Figure 4, which is a flowchart of call conference recording and takeover setting of the multimedia gateway management method of the present invention. During the call conference, when the first multimedia gateway is abnormal, the second multimedia gateway can be activated to reestablish the conference and take over without interrupting the call. The steps are described in detail as follows.

In step 410, the client starts a call session. During the process of establishing the call session, the traffic analysis and processing module 280 analyzes and processes all traffic packets and stores them in the database 260.

In step 420, the management and monitoring module 240 performs resource monitoring based on the current call traffic and the capacity of the multimedia gateway. If the multimedia gateway operates normally, it returns to step 410 to continue receiving new calls. If it is observed that the multimedia gateway in the call conference is abnormal due to insufficient resources, it enters step 430.

The above resources include the CPU computing time, memory storage space, and network I/O resources (such as communication ports) of the multimedia gateway. Insufficient network I/O resources will result in the inability to process incoming calls normally. Exhaustion of CPU computing and memory resources will cause intermittent voice due to lack of computing resources for voice encoding processing, affecting the call quality. The management and monitoring module 240 can monitor the resource usage of each multimedia gateway to determine whether each multimedia gateway has sufficient resources to process the currently ongoing call conversation.

In step 430, the management and monitoring module 240 allocates resources, i.e., adds and activates the second multimedia gateway and restores the call conference status. The call conference service stored in the database 260 in step 410 is restored by the service reorganization module 250.

In step 440, the second multimedia server will send a reINVITE/SDP or UPDATE/SDP packet to the client and the media user agent of the second multimedia gateway at the end of the process in Figure 3 to complete the takeover.

In step 450, the call session of the first multimedia gateway has been taken over by the second multimedia gateway, so the first multimedia gateway can release the resources of the call session and clear the data of the call session.

Please refer to Figure 5, which is a flow chart of the session reestablishment conflict setting of the multimedia gateway management method of the present invention. Since each multimedia gateway has its own independently established call session, it may also use the same RTP/RTCP communication port. In order to solve the problem that the multimedia gateway that takes over uses the same RTP/RTCP communication port when reestablishing the call session, the setting steps are as follows.

In step 510, during the conversation, the first multimedia gateway and the second multimedia gateway both use the same RTP/RTCP communication port to allow the client to call and talk with the media user agent of the multimedia gateway. Generally, RTP uses an even-numbered communication port, and RTCP uses the next port number of RTP, or uses the next communication port of RTP. For example, the communication port number used by RTP is port 20000, and the communication port number used by RTCP is port 20001.

In step 520, the second multimedia gateway prepares to take over the call conversation of the first multimedia gateway and restores the conversation information of the first multimedia gateway.

In step 540, when restoring the SDP information of the call session, the service reorganization module 250 checks whether the second multimedia gateway has occupied the RTP/RTCP communication port used by the call session of the newly taken over first multimedia gateway. If the service reorganization module 250 finds that the RTP/RTCP communication port has been occupied by the second multimedia gateway, it proceeds to step 530; if not, it proceeds to step 560.

In step 530, the service reorganization module 250 allocates a new communication port for use by the call conference newly taken over by the second multimedia gateway. The new communication port will not conflict with the existing call conference of the second multimedia gateway.

In step 550, the second multimedia gateway sends a reINVITE or UPDATE packet with updated SDP information (i.e., SDP field) to notify the client and the media user agent of the second multimedia gateway to switch the call session to the new communication port of the second multimedia gateway. The SDP field of the reINVITE or UPDATE packet contains the IP address of the second multimedia gateway and the port number of the new communication port. Finally, the call is taken over without interruption.

In step 560, if the RTP/RTCP communication port used by the newly taken over call session is not occupied, the second multimedia gateway sends a reINVITE or UPDATE packet to notify the client and the media user agent of the second multimedia gateway to switch the call session to the second multimedia gateway according to the normal mechanism. The SDP field of the reINVITE or UPDATE packet contains the IP address of the second multimedia gateway. Finally, the call is taken over without interruption.

Please refer to Figure 6, which is a flowchart of the non-sequential call conference reconstruction setting of the multimedia gateway management method of the present invention. It mainly does not need to reassemble in sequence according to the original packet sequence when restoring and reconstructing the conference packet message. In addition to increasing the efficiency of call reconstruction, it can also reduce the packets transmitted in the network. The steps are as follows.

In step 610, the high availability distribution module 270 starts to receive packet traffic of the call session. A complete and recoverable call session mainly includes three packets: INVITE, 200 OK, and ACK. The 100 Trying packet or 180 Ringing packet can be ignored and does not need to be processed specially. The session progress (183 Session Progress) packet carries SDP information, so its SDP content needs to be retained.

In step 630, the traffic analysis processing module 280 determines whether a farewell (BYE) packet is received; if a BYE packet is received, the process proceeds to step 620; if no BYE packet is received, the process proceeds to step 640.

In step 620, the traffic analysis processing module 280 deletes the relevant data and paired related information of the call conversation in the database 260 according to the information of the remote tag and local tag of the call conversation, and the process ends after completion.

In step 640, the traffic reorganization module 250 performs non-sequential packet processing. When restoring and reestablishing the session, the session is reestablished based on the remote tag and the local tag. However, it is possible that the client initiated the INVITE packet to establish the session initially. During the reestablishment, the media user agent of the multimedia gateway may simulate the INVITE packet. Therefore, the INVITE packet received by the client can be stored as the INVITE information or in the form of 200 OK. Conversely, the 200 OK packet responded by the media user agent of the multimedia gateway can be stored as the 200 OK packet information or in the form of INVITE packet.

In step 650, the traffic reorganization module 250 stores the INVITE packet, 200 OK packet of the local tag and the INVITE packet, 200 OK packet of the remote tag for each call used to reestablish the session in the database 260. Then, the non-sequential packet recovery setting is completed.

For example, when reestablishing a call conference, if a sequential reconstruction is adopted, it is necessary to send an analog INVITE packet from the input of the calling end to the called end in order to reestablish normally. If a non-sequential reconstruction is adopted, and an analog INVITE packet is sent from the called end, it will be found that the analog INVITE packet cannot be sent from the input of the called end to the calling end, but can only be sent from the output of the called end. In this regard, the dual form packet storage of step 640 can be adopted. According to this storage method, the input of the calling end and the output of the called end include not only the INVITE packet and the ACK packet, but also the 200 OK packet stored according to the converted form of the INVITE packet, and the output of the calling end and the input of the called end include not only the 200 OK packet, but also the INVITE packet and ACK packet stored according to the converted form of the 200 OK packet.

In this way, in addition to sending a simulated INVITE packet from the input pin of the calling end to the called end in accordance with the original packet sequence to reestablish the conversation, the simulation can also be initiated from the output pin of the calling end, or from the called end. Regardless of whether the simulation is initiated from the input pin or the output pin of the called end, the conversation can be reestablished.

For another example, if the calling end initiates a call to the called end, the packet records in the database 260 will be recorded from the calling end's input and output to the called end's output and input. If sequential restoration is adopted, it is necessary to restore from the calling end's input and output to the called end's output and input in the order of the records. If non-sequential restoration is adopted, it is not necessary to restore in this order.

Specifically, there are several ways to reconstruct the caller and the called party's conversation steps (entry/exit): (1) Reconstruct the caller's entry in the order of recording, then reconstruct its exit, then reconstruct the called party's exit, and then reconstruct its entry; (2) Reconstruct the called party's exit in the order of recording, then reconstruct its entry, then reconstruct the caller's entry, and then reconstruct its exit (the above two methods are equivalent and are in the same direction); (3) Non-sequentially reconstruct the caller's entry first, then reconstruct its exit, then reconstruct the called party's entry, and then reconstruct its exit; (4) Non-sequentially reconstruct the called party's entry first, then reconstruct its exit, then reconstruct the caller's entry, and then reconstruct its exit. (The above two methods are equivalent and both are in the same direction); (5) Non-sequentially, the outgoing foot of the calling end is first rebuilt, and then its incoming foot is rebuilt, and then the outgoing foot of the called end is rebuilt, and then its incoming foot is rebuilt; (6) Non-sequentially, the outgoing foot of the called end is first rebuilt, and then its incoming foot is rebuilt, and then the outgoing foot of the calling end is rebuilt, and then its incoming foot is rebuilt (the above two methods are equivalent and both are in the same direction); (7) Non-sequentially, the outgoing foot of the calling end is first rebuilt, and then its incoming foot is rebuilt, and then the incoming foot of the called end is rebuilt, and then the outgoing foot is rebuilt; (8) Non-sequentially, the incoming foot of the called end is first rebuilt, and then its outgoing foot is rebuilt, and then the outgoing foot of the calling end is rebuilt, and then its incoming foot is rebuilt (the above two methods are equivalent and both are in the same direction). The advantage of non-sequential reconstruction is that it does not need to be reconstructed sequentially from the perspective of the entire call conversation, but can be used to reconstruct the call conversation of the calling end or the called end individually from the perspective of the multimedia gateway. It can be reconstructed sequentially without following the original packet sequence. In this way, in addition to increasing the efficiency of call reconstruction, because all complex packets are simplified into three packets of INVITE/SDP, 200 OK/SDP and ACK, it can also reduce the packets transmitted on the network.

Please refer to Figure 7, which is a flowchart of the multimedia gateway automatic adjustment high availability continuous setting of the multimedia gateway management method of the present invention. It can actively add or reduce the system resources of the multimedia gateway according to the existing system resources and traffic flow to achieve system resource optimization. The steps are as follows.

In step 710, the management monitoring module 240 periodically monitors the resource usage of all multimedia gateways.

In step 720, the management monitoring module 240 detects whether there is any abnormality in the activity of the multimedia gateway; if the multimedia gateway is found to be abnormal due to a system failure or the system resource usage status is abnormal, then the process proceeds to step 740; if there is no abnormality, then the process proceeds to step 730.

The above-mentioned system resources may include the CPU computing time, memory storage space, and network I/O resources (such as communication ports) of the multimedia gateway. The management monitoring module 240 can monitor whether the system resources of all multimedia gateways under its jurisdiction are insufficient due to excessive number of call sessions, and check whether all multimedia gateways respond to messages normally through SIP queries. If the resources of the multimedia gateway are insufficient or the SIP query is not responded, the management monitoring module 240 determines that the activity of the multimedia gateway is abnormal.

In step 740, the management monitoring module 240 activates the new multimedia gateway, ready to take over the call session of the abnormal multimedia gateway.

In step 760, the abnormal multimedia gateway or the multimedia gateway that is requested to be shut down by the management monitoring module 240 checks whether the multimedia gateway itself has an ongoing call conversation that needs to be taken over by the newly activated multimedia gateway; if there is a call conversation that needs to be taken over, the process enters step 770; if not, the process enters step 780.

In step 770, the service recovery processor of the newly activated multimedia gateway performs the call non-sequential call negotiation reconstruction setup process and notifies the taken-over multimedia gateway that it can be shut down to reduce resource waste; after completion, enter step 780.

In step 730, the management monitoring module 240 determines whether it is necessary to reduce system resources. If the multimedia gateways are normal and the number of ongoing calls and conferences is small, it is necessary to reduce system resources and the process proceeds to step 750; otherwise, it proceeds to step 780.

In step 750, the management and monitoring module 240 may shut down redundant multimedia gateways. For example, the management and monitoring module 240 may shut down the multimedia gateway with the lowest number of call sessions to release resources. Then, the process proceeds to step 760.

In step 780, the multimedia gateway waits for the user to actively end the call, and then releases the call conference resources to end this process.

The present invention also provides a computer-readable medium, such as a centralized or distributed memory, floppy disk, hard disk or optical disk. The computer-readable medium is applied to a multimedia gateway system and stores instructions to execute the above-mentioned multimedia gateway management method.

When compared with other commonly used technologies, the technical solution of this invention has the following advantages:

The present invention has the technology of automatically adjusting resources. It does not need to deploy a large number of multimedia gateways. It can cope with short-term peak usage, so as to avoid wasting machine resources during off-peak usage. It can also effectively reduce the manpower of back-end maintenance and reduce the cost of machine equipment.

The present invention provides a call SIP packet analysis technology, which integrates and saves the packets of each initiated SIP conversation into a database, so as to provide an uninterrupted call function when restarting the service or switching to other multimedia gateways, and can seamlessly continue the call, thus improving the user experience.

The present invention provides a non-sequential recovery and reconstruction analysis technology for call SIP packets. If the client has gone through a complex call scenario, such as a three-party call, it is not necessary to follow the order of initiation by the calling and called ends. It can also directly simulate the INVITE packet initiated by the called end to reconstruct the conversation content. The conversation can be reconstructed more quickly and correctly, and the reconstruction is performed in the form of simulated packets without affecting the existing call status of the client.

The present invention provides a call conference takeover technology. When two multimedia gateways are activated at the same time, one of the multimedia gateways can take over the call conference of the other multimedia gateway, thereby achieving the call conference transfer function without interruption.

In summary, this invention is not only innovative in terms of technical ideas, but also has the above-mentioned multiple functions that are not available in the traditional methods used. It has fully met the statutory invention patent requirements of novelty and progress. Therefore, I have filed an application in accordance with the law and sincerely request your approval of this invention patent application to encourage inventions. I am grateful for your kindness.

110: Multimedia signal generation system

120:SIP application server

200: High availability multimedia gateway system

210~212: Multimedia gateway

220~222: Media User Agent

230~232: Service recovery processor

240: Management monitoring module

250: Communication service reorganization module

260: Database

270: High availability distribution module

280: Information analysis and processing module

Claims (9)

A high-availability multimedia gateway system includes: a plurality of multimedia gateways for providing a call conference function of a conference initiation protocol; a high-availability allocation module for allocating the call conference to a first multimedia gateway among the multimedia gateways when a user initiates a call conference; a traffic analysis and processing module for extracting establishment information of the call conference from a traffic packet of the call conference to store the establishment information; a management and monitoring module for monitoring whether the activity of the first multimedia gateway is abnormal during the call conference, so as to cause the first multimedia gateway among the multimedia gateways to be disconnected when the activity of the first multimedia gateway is abnormal. The second multimedia gateway takes over the call conversation; and a traffic reorganization module is used to generate a plurality of session reestablishment packets of the call conversation according to the establishment information, wherein the second multimedia gateway internally simulates the sending and receiving of the session reestablishment packets when taking over the call conversation, rather than actually sending and receiving the session reestablishment packets, so as to reestablish the state of the call conversation, and the second multimedia gateway sends a re-invitation packet or an update packet of the session initiation protocol to inform the client to switch the multimedia stream of the call conversation to the second multimedia gateway, so as to take over the call conversation while maintaining the call conversation uninterrupted. A high-availability multimedia gateway system as described in claim 1, wherein the high-availability allocation module allocates the call conference to the first multimedia gateway based on the resource usage. A high-availability multimedia gateway system as described in claim 1, wherein if the resources of the first multimedia gateway are insufficient to handle the call session, or the first multimedia gateway fails, the management monitoring module determines that the activity of the first multimedia gateway is abnormal. A high availability multimedia gateway system as described in claim 1, wherein the establishment information of the call conference includes the conference initiation protocol invitation packet of the caller end and the called end of the call conference, the conference initiation protocol field and the conference description protocol field of the conference initiation protocol acceptance packet and the conference initiation protocol confirmation packet, and the paired relationship between the caller end and the called end. A high-availability multimedia gateway system as described in claim 4, wherein the establishment information of the call session includes packets stored in mutually converted forms between the session initiation protocol invitation packet and the session initiation protocol acceptance packet, so that when the second multimedia gateway reestablishes the state of the call session based on the packets stored in mutually converted forms, it is not necessary to follow the original packet sequence of the call session. A high-availability multimedia gateway system as described in claim 1, wherein the management monitoring module is reused to monitor and provide resource usage of the multimedia gateways, and based on the resource usage, activate or deactivate at least one of the multimedia gateways to adjust the resource amount of the multimedia gateways. A high-availability multimedia gateway system as described in claim 1, wherein the traffic reorganization module is reused to check whether the communication port allocated by the first multimedia gateway to the call conference has been occupied by the second multimedia gateway when the second multimedia gateway takes over the call conference, wherein if the communication port has been occupied by the second multimedia gateway, the traffic reorganization module allocates a new communication port not occupied by the second multimedia gateway to the call conference after being taken over. A multimedia gateway management method includes: when a user terminal initiates a call session of a conference initiation protocol, the call session is assigned to a first multimedia gateway among a plurality of multimedia gateways; The establishment information of the call session is captured from the traffic packet of the call session to store the establishment information; during the call session, the activity of the first multimedia gateway is monitored for abnormalities, wherein when the activity of the first multimedia gateway is abnormal, the second multimedia gateway among the multimedia gateways is allowed to take over the call session; and according to the Establish information to generate multiple session reestablishment packets of the call session, wherein the second multimedia gateway internally simulates the sending and receiving of the session reestablishment packets when taking over the call session, rather than actually sending and receiving the session reestablishment packets, to reestablish the state of the call session, and the second multimedia gateway sends a re-invite packet or an update packet of the session initiation protocol to inform the client to switch the multimedia stream of the call session to the second multimedia gateway, so as to take over the call session without interrupting the call session. A computer-readable medium, used in a multimedia gateway system, stores instructions for executing the multimedia gateway management method as described in claim 8.

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