44 results about "Sip server" patented technology

The invention provides an implementation method for converting a GB28181 protocol RTP video stream into a low-delay HTTP-PS video stream, which is applied to an HTTP-PS server and comprises the following steps: step 1, receiving a streaming media playing HTTP connection request of a client; step 2, analyzing a device code DEV1 and a channel code CHANNEL1 of a target video source from the streaming media playing HTTP connection request, and creating an HTTP response object; step 3, randomly acquiring an unused TCP port P1 from the local machine, and starting the Socket service S1 to monitor the port P1; step 4, initiating a stream taking request to the SIP server to request the SIP server to send the RTP video stream of the target video source to the Socket service S1; step 5, after the Socket service S1 receives the RTP data after each PS data packet is subpackaged sent by the target video source, the RTP data is packed in real time to obtain the PS data packet, and the PS data packet is written into the HTTP response object, so that the client obtains the PS packet data from the HTTP response object and displays a frame of live broadcast picture; and step 6, when monitoring that the HTTP connection is disconnected, stopping the Socket service S1.

The invention provides an SIP (Session Initiation Protocol) authentication method based on a post quantum and national secret mixed signature algorithm, which is applied to an SIP client, and comprises the following steps: sending a first registration message to an SIP server, the first registration message comprising a target authentication control parameter, so that the SIP server generates a first reply message according to the target authentication control parameter; receiving a first reply message; obtaining a first mixed signature certificate of the SIP client according to the first reply message, generating a first random number, and performing mixed signature on the first random number to generate a first signature value; sending a second registration message to the SIP server, so that the SIP server verifies the first mixed signature certificate by adopting a pre-stored root certificate of the SIP client, and verifies the first signature value; and receiving second reply information, and indicating whether the SIP client is successfully verified or not. By integrating a post-quantum algorithm, SIP authentication has the characteristic of quantum attack resistance.

The embodiment of the invention relates to the field of data synchronization, and discloses a same-number multi-terminal call data synchronization method for a VoIP (Voice over Internet Protocol) system. In the invention, the existing SIP signaling channel between the terminals and the SIP server is multiplexed, and the control information is transmitted to negotiate to establish the end-to-end data channel which does not pass through the SIP server. And then, the terminal executes a set of four-stage handshake synchronization protocol based on an incremental change log through the end-to-end channel to synchronize call data. The problem that the related technology in the field depends on a synchronization server or is limited by near field communication is effectively solved, and real-time synchronization and consistent convergence of service data between terminals are realized on the premise of keeping the existing VoIP signaling architecture unchanged and not additionally deploying a special centralized data synchronization server; the system reliability is improved; and the deployment and maintenance cost is reduced.

The invention is suitable for the technical field of communication, and provides an SIP signaling transmission and flow control method and device and computer equipment, and the method comprises the steps: obtaining an SIP signaling flow prediction value in an SIP network through a flow prediction model, and triggering a flow adjustment strategy when the SIP signaling flow prediction value exceeds a threshold value; the SIP network comprises at least one SIP device and at least one SIP server, any SIP device monitors the state of itself in real time, if state change information needing to be synchronized appears, a difference data packet is generated according to the state change information, and the difference data packet is transmitted to the corresponding SIP server, so that the SIP server updates the stored state information of the SIP device according to the difference data packet; and otherwise, sending a heartbeat signaling to complete SIP periodic registration. On one hand, signaling accumulation and network congestion are effectively relieved through flow prediction and signaling priority distribution, and on the other hand, redundant data transmission is reduced through difference updating.

This application provides a method, apparatus, and computer-readable storage medium for making VoIP calls, relating to the field of communications. It can prevent the leakage of call-related information and improve the security of VoIP calls. The method is applied to VoIP and includes: obtaining a first set of call information and a second set of call information from the caller; the first set of call information includes a displayed number, a call prefix, and a called number; the second set of call information includes an identifier of the target communication line and an identifier of the target port of the target communication line; performing asymmetric encryption on the displayed number, call prefix, and called number to obtain target encrypted information; and sending first instruction information to a Session Initiation Protocol (SIP) server; the first instruction information includes the second set of call information and the target encrypted information, and is used to instruct the SIP server to make a VoIP call based on the second set of call information and the target encrypted information.

The invention provides a device of an SIP telephone terminal integrated with an SIP server, and the device comprises an SIP client module which comprises audio input and output equipment and is used for being responsible for a basic voice call function; the SIP server module is used for managing all registered SIP clients, processing routing, transferring, group calling and conference conversation of calls, and providing a voice mailbox and call records; the SIP relay module is used for converting or forwarding internal and external SIP signaling and media data; the network interface module is used for providing wired or wireless connection options to be connected to a local area network and the Internet and is responsible for data transmission; and the user management and configuration interface module is used for providing user-friendly interface configuration and various functions of management equipment. In this way, the SIP server function and the SIP telephone function are integrated into the single device, multiple functions are integrated into the single device, the hardware cost and the installation cost are reduced, and the stability of the communication system is enhanced.

The invention provides a communication method, system and device and a readable medium, and belongs to the technical field of communication. The method applied to a first intermediary server comprises the following steps: receiving a first initial session protocol (SIP) message from a target terminal, analyzing the first SIP message to determine a target SIP server, and forwarding the first SIP message to the target SIP server; and receiving a second SIP message from the target SIP server, analyzing the second SIP message to determine the target terminal, and forwarding the second SIP message to the target terminal. The method is used for improving the stability and reliability of SIP access service, and provides a basis for realizing stateless load balancing.

To enable even a user lacking in technical knowledge to easily introduce an advanced telephone function at low cost while utilizing an existing contract telephone number as is. [Solution] A program to be executed by an information processing device having a processor and memory, the program causes the processor to execute the following steps: acquire configuration information from a communication device having SIP server functionality installed at the user's base, the configuration information including the contract telephone number registered in the communication device, authentication information for the information processing device, and available extension numbers; use the acquired configuration information to register the information processing device as an extension terminal of the communication device based on SIP; and transfer calls addressed to the contract telephone number received by the information processing device as an extension terminal to the cloud PBX using a communication connection between the information processing device and the cloud PBX on the Internet.

This invention provides a method for converting GB28181 protocol RTP video streams into low-latency HTTP-PS video streams, applied to an HTTP-PS server, including the following steps: Step 1, receiving a streaming media playback HTTP connection request from a client; Step 2, parsing the target video source's device encoding (DEV1) and channel encoding (CHANNEL1) from the streaming media playback HTTP connection request, and creating an HTTP response object; Step 3, randomly obtaining an unused TCP port P1 from the local machine, and starting a Socket service S1 to listen on port P1; Step 4, initiating a stream retrieval request to a SIP server to request that it send the target video source's RTP video stream to the Socket service S1; Step 5, after receiving the RTP data after each PS data packet sent by the target video source, the Socket service S1 reassembles the RTP data into packets in real time to obtain PS data packets, and writes them into the HTTP response object so that the client can obtain PS packet data from the HTTP response object and display a frame of live video; Step 6, stopping the Socket service S1 when the HTTP connection is detected to be disconnected.

A device includes: a certificate check unit that, when a public key of a user A in a public key certificate stored in an IC card owned by the user A of a calling-side A telephone set using SIP is read and transmitted by the A telephone set and then received by a SIP server, checks validity of the public key certificate with an authentication server that authenticates the validity; a random number generation unit that generates a random number r when the validity is confirmed; and a signature verification unit that verifies that personal identity information on the user A in a DB is legitimate when a locked random number r, which is obtained by converting the random number r by using a private key kept in an IC card read by the A telephone set, is converted into the original random number r by using the public key.

A voice quality information collection device includes a SIP server log collection unit for collecting a SIP server log including an incoming call number, identification information of a destination carrier, and a line number, from a SIP server, an MG log collection unit for collecting an MG log including a TerminationID allocated by a predetermined logic and a delay time calculated by an RTCP report, from an MG, and a data generation unit for converting a line number included in the SIP server log into a TerminationID and integrating the post-conversion SIP server log and the MG log by using a TerminationID, to generate log integration information.

This application discloses a method and apparatus for transmitting VoIP media parameters. The method is applied to an initiating end, wherein the initiating end is one of an IAD and a SIP Server, and the corresponding receiving end is the other of the IAD and SIP Server. The method includes: after a normal VoIP call has been established between the IAD and the SIP Server, if it is detected that at least one of the packet duration and encoding / decoding rules in the current VoIP media parameters needs to be updated, a notification message is sent to the receiving end, wherein the notification message includes the value to be updated for the parameter currently to be updated; obtaining a response message from the receiving end to the notification message, wherein the response message carries a judgment result on whether the value to be updated for the parameter currently to be updated is supported; determining whether the parameter currently to be updated has been successfully updated based on the judgment result in the response message; and updating the VoIP media parameters using the value to be updated for the parameter currently to be updated after determining that the parameter has been successfully updated.

The invention discloses a cross-domain bridging method based on an SIP building intercom technology, and the method comprises the following steps: S1, enabling an indoor unit to be disposed at the edge of an internal and external network, enabling the internal network to be connected with an intercom system terminal device through a network, enabling the external network to be connected with a user network through WIFI, and registering a public network SIP server to communicate with an external network SIP device; and S2, the intranet entrance machine calls the indoor unit, and the indoor unit receives the session signaling and the media stream from the building intercom terminal. According to the invention, indoor unit SIP bridging is adopted as a transfer scheme, the compatibility and communication efficiency of the SIP building intercom system during cross-domain communication are improved, the problem of connection compatibility between different networks faced by the SIP building intercom system during cross-domain communication is solved, and meanwhile, the encryption technology in the data transmission process is matched, so that the communication efficiency is improved. The security and stability of data transmission are enhanced, and the use requirements are met.

The application provides a SIP session protection method based on conversation copying, comprising the following steps: collecting user SIP sessions of a user end SIP server, evaluating the quality of the user SIP sessions, and determining SIP session contents needing protection; copying the SIP session contents needing protection, and establishing a channel between SIP servers in conversation of the SIP session contents needing protection; constructing a defense mechanism on the channel, and protecting the SIP session contents needing protection based on the defense mechanism.

The application discloses an audio and video transcoding method, device and equipment and medium. The method is executed by a SIP server, the SIP server comprises a plug-in media board, at least one plug-in transcoding component is configured on the media board, and the method comprises the following steps: acquiring a source audio and video format of a source SIP terminal initiating an audio and video call and a target audio and video format of a target SIP terminal answering the audio and video call; when the source audio and video format is inconsistent with the target audio and video format, locating a source transcoding component matched with the source audio and video format and a target transcoding component matched with the target audio and video format in each media board; and performing bidirectional transcoding on audio and video content interacted between the source SIP terminal and the target SIP terminal by calling the source transcoding component and the target transcoding component. The technical scheme of the application solves the problem of fixed transcoding capacity of an existing integrated SIP server, and avoids the influence of damage of a single transcoding component on the overall function of the SIP server.

The invention discloses a VOIP (Voice Over Internet Protocol) connection interaction method of ONU (Optical Network Unit) equipment based on an SIP (Session Initiation Protocol). The VOIP connection interaction method comprises the following steps of S1, preparing pre-connected message content in the S2, the ONU equipment sends a pre-connection request message to the SIP server; s3, after the SIP server receives the pre-connection request message, field information supported by the server is put in a response message, and the response message is sent to the ONU equipment; s4, after the ONU equipment receives the response message, the field information is analyzed, and the specific format of each field is determined; and S5, the ONU equipment reconstructs a normal SIP request message. The invention is based on the SIP protocol, is used for adapting to the VOIP voice function of the ONU equipment, provides a pre-connection method for the SIP registration function of the ONU equipment and other subsequent operations, and provides a more convenient, stable and efficient way for adapting to the VOIP and a customer service provider.

The application relates to the technical field of communication performance testing, and provides a server media stress testing method and system, electronic equipment and a storage medium. An implementation scheme is as follows: a calling SIPp and a called SIPp send a start testing request to a sending party testing media server and a receiving party testing media server by calling an external interaction interface, the start testing request carries respective local media parameters, opposite media parameters and a session identifier; the sending party testing media server generates testing information in response to the start testing request, encodes the testing information and media data, obtains a media data packet, and enables the SIP server to send the media data packet to the receiving party testing media server in the form of an RTP stream; the receiving party testing media server receives and analyzes the media data packet, obtains analysis data, tests the RTP stream based on the analysis data, and generates a media stress testing result. The application can improve testing granularity and testing quality.

According to the installation and application of the uniapp plug-in of the communication technology in the coal mine industry based on the sip protocol, a corresponding project is created or opened in HBuilder X, the communication module plug-in is checked in app native plug-in configuration of project mangist, a user-defined base is packaged, the project for loading the communication module plug-in is selected, the user-defined base is obtained, then the user-defined base is selected when the project is operated, and the user-defined base is used for installing the uniapp plug-in of the communication technology in the coal mine industry according to the installation and application of the uniapp plug-in of the communication technology in the coal mine industry. An SIP server is deployed on the ground or under the ground to serve as a core node of communication, and projects loaded with the communication module plug-ins are communicated through the SIP server. According to the coal mine industry communication technology uniapp plug-in based on the sip protocol, the high efficiency and stability of audio and video calls are ensured through the optimized coding and decoding technology and the SIP protocol, and the safety of communication content is ensured by adopting the advanced encryption technology. The plug-in is developed based on a UniApp framework, supports multi-platform deployment, is suitable for different types of mobile devices, provides abundant APIs and event monitoring mechanisms, and is convenient for developers to quickly integrate and use.

The invention discloses an encrypted video and video key encryption key synchronous transmission method and a related device thereof, and the method comprises the steps: after an SIP server receives an Invite request which is sent by a media stream receiver and carries an SDP message, the connection between a media server and a media stream sender is established through three-party call control; establishing a connection between a media stream receiver and a media server through a B2BUA proxy mode; and when the VKEK is changed, the SIP server sends a message to notify the media server, the changed VKEK is issued to each path of encrypted video distribution link by the media server, then the changed VKEK is packaged into an RTP data packet by each link, and the RTP data packet is sent to a media stream receiver before the next key frame is sent to decrypt the video. The problem that the streaming media VKEK and the encrypted video stream are not synchronous is solved, meanwhile, the development difficulty of media management and the client side is reduced, and popularization is facilitated.

To provide a communication system that contributes to enabling direct IP telephone communication between HGWs via the Internet without continuous intervention of an IP telephone network or a SIP server.SOLUTION: A communication system includes a first home gateway and a second home gateway that are connected via an Internet network. The first home gateway acquires second global IP address information of the second home gateway. The second home gateway acquires first global IP address information of the first home gateway. The first home gateway and the second home gateway use the acquired first global IP address information and second global IP address information to establish a voice call via the Internet network.SELECTED DRAWING: Figure 1

1. The name of the product under this design: Smart Intercom (A16V Series). 2. Purpose of this product: Used with SIP server for intercom calls, broadcast notifications, background music, video surveillance, etc. 3. The design key point of this appearance design product lies in the shape. 4. The picture or photo that best illustrates the design points: three-dimensional picture.

The present invention discloses a video stream transmission method and device, which includes: establishing a SIP communication channel based on the WebSocket protocol; using the SIP communication channel to forward a SIP server registration request containing user information to a SIP server via a GB access platform; obtaining the video code of a target video acquisition device; selecting a video stream channel type to establish with a streaming media server based on the video code, browser version, and decoder support status of the target video acquisition device; creating a first video stream channel with the streaming media server through a first SDP negotiation with the streaming media server, or creating a second video stream channel with the streaming media server through a second SDP negotiation with the streaming media server, based on the selected video stream channel type; and receiving a video stream from the target video acquisition device sent by the streaming media server. Based on the WebRTC compatible GB / T28181 protocol framework, the method implements video stream transmission and playback in both H.264 and H.265 encoding formats.

The application discloses a kind of GoIP equipment verification method, equipment, device and storage medium.The application discloses: each IP address in the region to be scanned is sent SIP protocol request, each IP address is screened according to the response result of each IP address, determine the IP address to be verified in each IP address, obtain the response message sent based on SIP protocol request of IP address to be verified, whether the device corresponding to IP address to be verified is GoIP equipment according to response message verification IP address to be verified;Since the application sends SIP protocol request to each IP address in the region to be scanned, since GoIP equipment is communicated between SIP protocol and SIP server, so it can be screened according to the response result of each IP address IP address to be verified, according to the response message sent by IP address to be verified, IP address to be verified is verified, compared with prior art verification method, the application can effectively and accurately identify GoIP equipment, and can accurately track and locate GoIP equipment based on IP address.

To automatically obtain a relocation destination number for a fixed-line telephone connected to an IP network.SOLUTION: A telephone number detection device 1 is configured by a computer system and includes a control unit 11, a memory unit 12 that stores telephone numbers, and a communication interface 13b that mediates data communication between the control unit and an IP network 100. When a code corresponding to the relocation of a destination number is sent from a SIP server 103 at the time of sending a call whose destination number is a telephone number to be investigated and that is supposed not to be received to the SIP server, the control unit receives and records the audio data of relocation guidance sent from within the IP network by sending a call for voice communication with the destination number to the SIP server, analyzes the recorded audio data to identify a relocation destination number included in the relocation guidance, and stores the relocation destination number in the memory unit.SELECTED DRAWING: Figure 3

The invention relates to the technical field of DECT, and discloses a communication method and communication equipment based on DECT. The method comprises the steps that after a base station mode is started, a registration request is sent to an SIP server through a wireless local area network module; under the condition that the registration request is successfully responded, SIP accounts are distributed to the terminal and other mobile terminals registered to the terminal through a DECT module; and communication is established between other mobile terminals registered to the terminal and the SIP server according to the SIP account through the DECT module and the wireless local area network module on the communication connection established with the SIP server and the other mobile terminals registered to the terminal. According to the invention, the dependence on the base station is overcome, the number of network devices needing to be deployed is reduced, the network management cost is reduced, the cost of purchasing the devices is reduced, the limitation of the communication range is broken through the mobility of the mobile terminal providing the function of the base station, the communication is more flexible, and the application is wider.

The application discloses a kind of based on unified port's SIP and RTP tunnel encryption forwarding method and system.The method is cooperated by tunnel client deployed in client side and tunnel server side of server side, including: based on unified port, establishing encrypted communication tunnel between the two;Tunnel client receives SIP signaling, modifies media connection information in it, and then sends to server side through tunnel encryption, and is forwarded to SIP server by server side;Tunnel client receives RTP media stream, and then sends to server side through tunnel encryption after additional header containing forwarding destination information is encapsulated, and is forwarded to SIP server according to header information by server side.The application tunnels and encrypts transmission of SIP signaling and RTP media stream through single port, effectively avoids network protocol ban, greatly simplifies firewall configuration, significantly improves NAT penetration success rate and communication security.

The invention provides an IMS (IP Multimedia Subsystem) multimedia call method and system. The method comprises the following steps: calling equipment initiates a session request to an IMS SIP server platform, wherein the session request comprises type information of the calling equipment; wherein the calling device type information at least comprises a calling type; the IMS SIP server platform identifies the call type; when the call type is an internal call, the IMS SIP server platform preferentially inquires whether a called device and the calling device are registered on the same IMS SIP server platform or not inside the IMS SIP server platform; if the session request and the media stream are registered on the same IMS SIP server platform, entering an internal calling mode, that is, the session request and the media stream are internally circulated through the IMS SIP server platform; and if the session request and the media stream are not registered on the same IMS SIP server platform, entering an outbound mode, that is, the session request and the media stream are transferred to an operator group IMS platform for addressing calling.

The invention belongs to the technical field of digital information transmission, and relates to a broadcasting method and system based on an SIP (Session Initiation Protocol) and an RTP (Real-time Transport Protocol). In the broadcasting method, at least one multicast address is established and configured for a terminal group when no broadcasting task exists, and terminal equipment in the terminal group is added into all multicast addresses; searching a target terminal group where a target terminal is located when a broadcast task exists, and selecting a target multicast address; sending a multicast quit command to a non-target terminal in the target multicast address, and sending a push stream to the target multicast address through an RTP protocol; and after the broadcast task is finished, adding the non-target terminal into the target multicast address. The broadcasting system comprises a dispatching host connected with an SIP server and a core switch, and further comprises an underground looped network, a plurality of looped network switches form annular connection through the underground looped network, one looped network switch is connected with the core switch, and other looped network switches are connected with terminal equipment. According to the invention, broadcast delay and broadcast missing are avoided, and the answering reliability of the terminal equipment is improved.

The invention discloses a voice short number mutual dialing method and system based on an enterprise gateway, the gateway and a medium, and relates to the technical field of enterprise voice communication. The method comprises the following steps: a short number configuration process: configuring a unique short number for each telephone, and establishing a corresponding relationship between each short number and each telephone to form a short number list; a call control process: when it is detected that the first phone dials the short number in the short number list, a second phone in the short number list is matched according to the short number, and a ringing signal is sent to a virtual voice port of the second phone; and when an off-hook event of the second telephone is detected, modifying a mapping relationship between the physical voice ports and the virtual voice ports of the first telephone and the second telephone, and performing input stream redirection on the virtual voice ports of the first telephone and the second telephone, thereby realizing conversation between different telephones under the same gateway. According to the invention, the short number mutual dialing conversation between the telephones under one gateway is realized without depending on an external PBX / SIP server, so that the internal communication cost of an enterprise is reduced, and the reliability is high.

The present disclosure provides a communication method and system, a device, and a readable medium. The communication method applied to a first intermediate server comprises: receiving a first SIP message from a target terminal, parsing the first SIP message to determine a target SIP server, and forwarding the first SIP message to the target SIP server; and receiving a second SIP message from the target SIP server, parsing the second SIP message to determine the target terminal, and forwarding the second SIP message to the target terminal.