25 results about "Sip server" patented technology

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.

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.

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.

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 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 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.

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.

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 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 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 invention relates to a soft switch double-center management system and management method, and relates to the technical field of voice soft switch communication, and the method comprises the steps that a site management module configures and provides site information of a plurality of site devices, and the site information comprises site names and geographical location identifiers; the center management module configures and provides double-center information including name and service parameter setting; the main and standby routing configuration module configures a main and standby routing strategy, a control signaling and a media stream outlet path in the SIP server based on center and site information; the ACD queuing distribution module configures a seat distribution strategy based on station and center information; and the RTC gateway configuration module configures the butt joint relation of each center internal device. According to the application, when a single center fails, the other center takes over all services, so that the reliability, the flexibility and the resource utilization efficiency of the communication system are effectively improved, and the requirements of the communication services on service continuity and the requirements of rapid disaster recovery, dynamic load balancing and unified operation management are met.

A session initiation protocol (SIP) server facilitates defense against denial of service attacks. It receives a request from a transmitting SIP endpoint in a communication network, such as a 5G communication network, to establish a communication session with a destination SIP endpoint in the communication network; calculating a token from the encrypted secret and the requested attribute; the request is dropped and a challenge message is formed that includes the token. The SIP server sends a challenge message to the sender to trigger a second request from the sender including the token.

The invention relates to an SIP (Session Initiation Protocol) rapid re-registration method and system based on WAPI (Wireless Local Area Network Authentication and Privacy Infrastructure) secure access, and the method comprises the steps: collecting access identification information and current network state parameters of terminal equipment, completing identity authentication operation on the terminal equipment based on a WAPI authentication mechanism, and generating an authentication state object according to an authentication result; carrying out encryption binding on the authentication state object and access identification information of the terminal equipment, and writing an encryption binding result into an authentication state cache structure to form an authentication state cache; the method comprises the following steps: sending an SIPREGISTER request to an SIP server, after receiving an SIPREGISTER response message, extracting registration parameter data from the SIP registration response message, and generating a registration state snapshot object; and carrying out association binding on the registration state snapshot object and the access identification information of the terminal equipment to form a registration snapshot mapping structure. The method and the device have the effect of enhancing the continuity of terminal access.

This invention belongs to the field of building intercom systems, specifically a method for an indoor unit to forward network camera video to a mobile app. The method includes the following steps: S1: The indoor unit locates a network camera on the same network as the indoor unit and adds it to the indoor unit's network camera list; S2: The indoor unit registers with the SIP server using account sip1; S3: The mobile app also registers with the SIP server using account sip2. This invention uses the indoor unit's RTSP to retrieve network camera data and forward it to the mobile app. It can monitor both the indoor unit and the network camera, and allows for real-time switching, effectively solving the problem that residents' mobile apps cannot monitor network cameras in public areas of the community, and also resolving the issue of residents needing to install different manufacturers' apps when using different brands of network cameras.

A serverless Page Party (SPP) station is provided for page announcing and party line conferencing that is configured to implement mutual provisioning to allow all SPP stations within a system to mutually maintain their configurations in runtime, and multiple master station negotiation and master failover. Plural SPP stations are connected with Ethernet cable or WIFi to a network and therefore without need for expensive and cumbersome cabling and daisy-chain configuration among stations. SPP stations route packets between each other via multicast technology and without need for a IP-PBX or other server (e.g., a SIP server) for controlling inter-station connections. In Mutual Provisioning Mode, SPP stations operate using a system configuration obtained from other SPP stations already on the network using a command channel. System configuration is maintained by a SPP station designated as a master station. Plural SPP stations can negotiate among themselves to designate a new master station.

A session initiation protocol (SIP) server contributes to defending against denial of service attacks. It receives, from a sending SIP endpoint in a communications network, such as a 5G communications network, a request to establish a communications session with a destination SIP endpoint in the communications network; computes a token from a cryptographic secret and a property of the request; drops the request and forms a challenge message comprising the token. The SIP server sends the challenge message to the sender to trigger a second request from the sender comprising the token.

The invention provides a teleconference processing method and device, an SIP server and a storage medium, and relates to the field of voice communication. The method comprises the following steps: an SIP server establishes conference connection and generates a list of participants according to received conference joining requests initiated by a plurality of terminals; sending a preset prompt tone to all target terminals in the participant list to trigger a voice response of each target terminal, and taking the target terminal returning the verification voice stream as a to-be-discriminated terminal; performing voice recognition on the verification voice stream returned by each to-be-discriminated terminal to obtain an uploading recognition text corresponding to each to-be-discriminated terminal; and the to-be-processed terminal with the abnormal uploaded voice is determined based on the uploaded recognition text corresponding to each to-be-discriminated terminal, so that participation quality discrimination can be completed without manual intervention, the autonomous operation and maintenance capability and the user experience of a conference system are improved, and the method is suitable for high-reliability communication scenes such as unattended operation, remote cooperation and emergency scheduling.

The application discloses a communication penetration method and system based on a session initiation protocol, and the method comprises the following steps: forcibly intercepting a data packet sent by a SIP terminal to a SIP server, converting a source IP of the data packet into a public network IP, converting a source port of the data packet into a target port, replacing a private IP and a port carried in a header connection field in SIP signaling of the data packet with the public network IP and the target port, and replacing a private media IP and a port in SDP information with a public network IP and a target media port when the SIP signaling carries the SDP information. Therefore, it can be seen that a boundary device uniformly intercepts and processes SIP data packets, fixes a port conversion relationship during registration, synchronously modifies a private address and a port in a signaling header into boundary device public network information, realizes consistency of network layer and application layer address conversion, and avoids SIP signaling address mismatch in a NAT environment.

The present application belongs to the technical field of digital information transmission, and relates to a broadcasting method and system based on SIP protocol and RTP protocol. In the broadcasting method, at least one multicast address is established and configured for a terminal group when there is no broadcasting task, and terminal devices in the terminal group are added to all multicast addresses; when there is a broadcasting task, a target terminal group where a target terminal is located is found, a target multicast address is selected, a non-target terminal in the target multicast address is sent an exit multicast command, and a push stream is sent to the target multicast address through the RTP protocol; and after the broadcasting task is completed, the non-target terminal is added to the target multicast address. The broadcasting system comprises a dispatch host connected with a SIP server and a core switch, and further comprises an underground ring network, a plurality of ring network switches form a ring connection through the underground ring network, one ring network switch is connected with the core switch, and the other ring network switches are connected with terminal devices. The present application avoids broadcasting delay and broadcasting missed broadcast, and improves the call reliability of the terminal devices.