37 results about "Call establishment" patented technology

Disclosed are systems and methods for display confirmation of verified calling party attributes at the time of call establishment. A call initiation request corresponding to a calling party is received and includes a set of verified calling party attributes. A subset of the verified calling party attributes is provided for display on a communication device associated with a call recipient identified by the call initiation request and in response, a display acknowledgement is obtained, comprising a listing of verified calling party attributes recorded as being displayed during establishment of a communication session between the calling party and the call recipient. The display acknowledgement is validated and a display confirmation is generated to indicate particular verified calling party attributes that were successfully displayed during establishment of the communication session.

The invention provides an IMS voice calling method and system, a medium and a product, and relates to the technical field of communication, the method is applied to an IMS network architecture, and the IMS network architecture comprises a first network element, a second network element, a third network element and a fourth network element. The method comprises the following steps: performing resource use prediction based on historical data and network state information through a first network element collaborative proxy call session control function (P-CSCF); performing network evaluation through the second network element to obtain network state information; performing session parameter caching and preloading of the terminal through cooperation of the third network element and a serving call session control function (S-CSCF); and caching the call related data of the terminal by cooperating the fourth network element with a home subscriber server (HSS) and a query call session control function (I-CSCF). According to the invention, call establishment time can be reduced.

The invention discloses a voice communication method and device, terminal equipment and a computer readable storage medium, and the method comprises the steps: obtaining the information of a target cell corresponding to a voice communication request when the voice communication request is received; determining support state information of the target cell for the first call mode according to the information of the target cell; and when the support state information is first state information, establishing a voice call in a second call mode, the first state information representing that the target cell does not support establishment of the voice call in the first call mode. By adopting the method provided by the invention, whether the target cell supports the establishment of the voice call in the first call mode or not can be predicted to obtain the support state information, and if it is predicted that the target cell does not support the establishment of the voice call in the first call mode, the voice call is directly established in the second call mode. Repeated attempts to establish the voice call in the first call mode are not needed, the voice call establishment time is saved, and the user experience is improved.

The application relates to an emergency call method, system and device, electronic equipment and a storage medium. The method comprises the following steps: in response to an emergency call trigger event, generating a call message and starting a first redial timer; the call message is used to initiate an emergency call with a public safety answering point; the first redial timer is configured with a first redial duration; a call establishment response of the emergency call is acquired; if the call establishment response indicates that the communication establishment fails, the emergency call is redialed according to a preset redial rule within the first redial duration, so that the emergency call is successfully established. The method can initiate an emergency call in response to an emergency call trigger event, and if the call establishment response indicates that the communication establishment with the public safety answering point fails, the emergency call is redialed according to the preset redial rule, so that the emergency call between the vehicle-mounted system and the public safety answering point is automatically established, and the safety performance of the vehicle-mounted system is improved.

The present application relates to the technical field of cluster communication, and proposes an intelligent fusion cluster group call method, steps of which include: configuring group members and member types thereof, wherein the member types at least include a cluster terminal, a dispatch station, and a non-cluster terminal, wherein the non-cluster terminal includes at least one of a traditional fixed phone, a mobile phone and a smart camera; a cluster core network performs corresponding media adaptation operation and right adaptation operation on different types of group members in a group call establishment process and a group call process according to the pre-configured member types, so that the non-cluster terminal can access and participate in the cluster group call service. The present application enables the public network terminal and the intelligent device originally not supporting the group call and the right operation to be integrated into the cluster communication process, expands the terminal compatibility and the service coverage range of the cluster system, and meanwhile maintains the necessary media control and right management characteristics in the cluster dispatch.

The invention discloses a call establishment method and network side equipment, and belongs to the technical field of communication, and the call establishment method comprises the steps that a first network element requests a first transmission address from an AGW deployed on a satellite under the condition of determining that a first condition is met, and the first network element provides service for first UE; the first network element sends the first transmission address to a second network element; the first condition comprises at least one of the following conditions: the first UE and the second UE can perform UE-SAT-UE communication; the first UE performs SDP negotiation of an early session or an early media; the AGW deployed on the satellite can communicate with the IMS user plane network element deployed on the ground; the first UE is not roaming; the satellite accessed by the first UE is a satellite in a regeneration mode; receiving a satellite identifier of the second UE; wherein the first UE is calling party UE, and the second UE is called party UE.

Provided are a wireless communication method, a terminal device, and a network device. The method comprises: a terminal device sending first indication information to a network device, wherein the first indication information is used for indicating whether the terminal device performs a voice call by means of a first radio access type. By means of first indication information, a network device can be aware of whether a terminal device performs a voice call by means of a first radio access type, thereby executing a corresponding voice call establishment process.

The invention belongs to the technical field of wireless communication, discloses a VoIP terminal active roaming method based on transmission layer two-dimensional monitoring, and belongs to the technical field of wireless communication. According to the method, after a VoIP call is established, an independent monitoring task is started in a user space, and a sending error counter and the total length of a sending queue of a transmission layer are collected at a fixed period by reading a network statistics interface provided by an operating system; calculating a sending error increment in each sampling period, and executing two-dimensional congestion judgment: if the sending error increment exceeds a first threshold value or the total length of the sending queue exceeds a second threshold value, marking as a suspected congestion state; and carrying out congestion confirmation and recovery judgment by adopting an asymmetric time filtering mechanism. According to the invention, active roaming triggering is realized by directly monitoring the queue state of the transmission layer in the user space, the method does not need to modify a system kernel or an application program, can respond immediately at the initial stage of link congestion, and provides smoother and more stable VoIP call experience for the user.

Systems and techniques are described for establishing one or more virtual sessions between users. For instance, a first device can transmit, to a second device, a call establishment request for a virtual representation call for a virtual session and can receive, from the second device, a call acceptance indicating acceptance of the call establishment request. The first device can transmit, to the second device, first mesh information for a first virtual representation of a first user of the first device and first mesh animation parameters for the first virtual representation. The first device can receive, from the second device, second mesh information for a second virtual representation of a second user of the second device and second mesh animation parameters for the second virtual representation. The first device can generate, based on the second mesh information and the second mesh animation parameters, the second virtual representation of the second user.

The embodiment of the invention provides a communication method, and the method can be applied to a registration scene or a call establishment scene of an internet protocol multimedia subsystem (IMS). In the method, the first IMS network element controls the operation related to the call of the first terminal device through the second duration, and the paging duration, the warning mechanism, the network access mode, the satellite alignment duration and the like are considered when the second duration is determined, so that the problem that the call is mistakenly terminated due to the fact that the fixed value of the timer is smaller than the paging duration can be reduced. And invalid waiting of the calling terminal due to the fact that the fixed value of the timer is far larger than the paging duration can be reduced.

This application discloses an emergency call control method and electronic device, relating to the field of terminal technology. The emergency call control method includes: a terminal device initiating an emergency call in response to an emergency call triggering operation; the terminal device receiving first indication information sent by a network device; if the first indication information indicates that the emergency call establishment has failed, the terminal device collecting distress voice data and re-initiating the emergency call; and the terminal device sending distress voice data after detecting that the re-initiated emergency call has been successfully established. This technical solution can improve the success rate of emergency calls, prevent users from missing valuable opportunities for help, and better protect user safety.

The voice call establishment method comprises the following steps: receiving a first call request of a first client; selecting a first intermediate number from a number pool of a first client according to the first call request; and establishing voice communication connection with the first client and the client called by the first client based on the first intermediate number. According to the method, when the client initiates the call request to request the voice call, one target intermediate number can be selected from the plurality of intermediate numbers in the bound number pool to establish the voice call, so that the client is prevented from being bound with a single intermediate number, the utilization rate of the intermediate number is improved, and the voice call efficiency is improved. And the service efficiency and the satisfaction degree of the voice communication of the client are improved.

The application discloses a kind of RTP message replay test method and device based on timing keeping, it is related to RTP message replay technical field.The method includes: obtaining message file, configuration test network card address, the address of device under test, and filter out RTP message to be replayed;With the SIP interaction of device under test, complete call establishment, media parameter negotiation, and obtain the session parameter of this session;The relative arrival time interval of each original message in RTP message to be replayed and RTP time stamp increment are parsed, and the message sending rate is controlled according to the relative time interval of original message when replaying;The specified field in replayed RTP message is dynamically revised to be consistent with the session parameter obtained by SIP signaling call process negotiation, so that device under test identifies replay stream as the legal media stream of current call.The RTP message replay test method and device based on timing keeping provided by the application realize quickly, accurately reproduce and locate the compatibility and network abnormality of video conference equipment.

The invention discloses a processing method of playback services, a satellite voice access gateway and home network equipment. The method comprises the following steps: acquiring a radio access technology (RAT) type of an access terminal, the access terminal comprising a calling terminal initiating a call or a called terminal responding to the call; under the condition that the RAT type is a first RAT type, generating playback suppression information, the first RAT type including non-ground network access, and the playback suppression information being used for indicating suppression of playback type services; and sending the playback suppression information to the home network of the calling terminal and / or the home network of the called terminal. According to the method and the device, the technical problems that the call establishment time is too long due to extra message interaction in bandwidth limited scenes such as a satellite communication network, or sound quality codes of playback services cannot meet transmission requirements, so that playback contents cannot be normally played, and the like are solved.

The application discloses a kind of fusion communication service management method, system, equipment and medium, cover from data acquisition, intention generation, service flow matching, user confirmation, the whole-process closed loop of adaptive optimization of execution, realize from listen to do one-stop service, without user jumps out call interface, in call establishment or in the middle of real-time data acquisition and execution service, realize the accompanying fusion of business handling and voice call.

Embodiments of the present application provide a communication method. The method can be applied to an Internet Protocol Multimedia Subsystem (IMS) registration scenario or a call establishment scenario. In the method, a first IMS network element controls, by means of a second duration, a call-related operation of a first terminal device. In addition, taking into consideration a paging duration, an alert mechanism, a network access mode, a satellite alignment duration, etc. while determining the second duration can mitigate the problem that a call is erroneously terminated due to a fixed value of a timer being less than the paging duration, and can also reduce meaningless waiting by a calling terminal due to the fixed value of the timer being much greater than the paging duration.

The invention provides a video call establishment method and device, equipment, a medium and a program product, and relates to the technical field of communication. The method is applied to a calling terminal and comprises the steps that a call request is sent to a called terminal through an IMS network, and the call request carries a first holographic communication identifier representing that the calling terminal has the holographic communication capability and information of a plurality of first transmission channels used for receiving multiple paths of video streams of the called terminal; receiving a response message sent by the called terminal through the IMS network; and establishing a video call with the called terminal based on the response message. By adopting the method provided by the invention, the terminal not only can carry out holographic video call under the IMS network, but also can carry out single-path video call under the IMS network, so that the video call capability of the terminal under the IMS network can be greatly improved; the problem that the requirement of a user for an immersive holographic video call cannot be met when a terminal in an IMS network only adopts a single camera to collect a single-path video stream in the prior art is solved.

The invention discloses a public network talkback method, a server, a storage medium and a program product, and relates to the technical field of talkback. In the method, when a server sends a wake-up message through a system push channel to start public network talkback application software in a target client, multi-port detection is initiated to target client equipment in parallel by using a pre-acquired historical media transmission port set; therefore, the detection process of the availability of the port and the process of waking up the public network talkback application software in the target client and returning the connection request are synchronously performed, and the total time of call establishment is compressed. Moreover, by generating the rapid media negotiation parameter and issuing the rapid media negotiation parameter together with the determined emergency communication port, multiple rounds of time-consuming interaction in a standard media negotiation process are skipped, and serial steps and network round-trip times in a call establishment process are reduced, so that the overall delay of background calls is shortened, and the instantaneity of public network talkback communication is improved.

A call establishment method includes: in a case that a first INVITE request sent by a second electronic device is received and that an INVITE ACK message sent by the second electronic device is not received, receiving a second INVITE request; and in a case that it is determined that both the second INVITE request and the first INVITE request are sent by the second electronic device, sending a target message to the second electronic device, and establishing a communication connection based on the second INVITE request. The target message is used for interrupting a communication connection established on the basis of the first INVITE request.

The application relates to an access gateway modification method and device, communication equipment, a storage medium and a computer program product. The method comprises: performing modification of an access gateway (AGW) in a multimedia subsystem (IMS) session establishment process. The method can effectively save signaling overhead in an IMS voice session establishment process of a user, shorten the call establishment delay, and ensure that limited on-board resources are effectively utilized, so that the network can maintain high efficiency allocation when more heterogeneous resources are available.

The invention relates to the technical field of communication security and data processing, and discloses an illegal call real-time truncation method and system based on AI analysis, and the method comprises the steps: capturing a bidirectional audio stream after a call is established, and carrying out the preprocessing of the bidirectional audio stream; generating a text stream by using streaming speech recognition; semantic, acoustic and dialogue state features are extracted in parallel, a real-time risk score is calculated through multi-modal fusion, and a risk level is judged. Performing hierarchical response according to the risk level; performing non-blocking intervention of vision, touch and audio mixing when the risk is medium or high; and when the risk is extremely high and the duration exceeds a threshold value, sending an instruction through a communication bottom layer to forcibly cut off the call. According to the method, semantic comprehension and acoustic feature analysis are combined, dialogue state tracking is introduced, the false alarm rate is effectively reduced, and accurate protection of fraud calls is achieved through bottom layer physical truncation.

The application belongs to the technical field of audio and video communication fault recovery and session maintenance, and relates to an audio and video call recovery method and system based on memory cache. The method comprises the following steps: configuring three servers, wherein the first two servers are deployed with signaling services, media services, cache nodes and node monitoring agents, and the third server is deployed with a node monitoring agent; during an audio and video call establishment process, signaling context is written into a master node in real time, media channel metadata is written into the master node in real time, and the data written into the master node is copied to a slave node; when it is determined that the master node is unavailable, a node monitoring agent cluster triggers master-slave node switching, and the cache node of the second server is selected as a new master node; and audio and video calls are recovered based on the current signaling context and the current media channel metadata on the second server. The application can automatically recover calls within 3 seconds under a low-cost architecture, and users do not need to perform any operation.

Method for establishing a multimedia call between a caller device and a called device over a telecommunication network, comprising enabling access (S6) to at least one application provided by a data channel server (DCS) from at least one device (UE) between said caller device and called device, whereas said at least one application belongs to a set of recommended applications selected (S3) for said call according to contextual information related to said at least one device.

The invention discloses a network telephone signaling interaction method, device and equipment and a storage medium, and relates to the technical field of communication interaction, and the network telephone signaling interaction method comprises the following steps: receiving a call establishment request from a calling party client to a called party client, the call establishment request comprises a called party identifier and geographical location information of a calling party; determining a target node candidate set according to the called party identifier and a preset mapping table, and determining a target node in the target node candidate set according to the geographical location information of the calling party; acquiring a communication address of the called party client from the target node; and establishing a signaling interaction link between the calling party client and the called party client according to the communication address so as to realize signaling interaction. According to the invention, the transmission delay of network telephone signaling interaction can be reduced.

The present invention relates to the technical field of communications, and in particular to a communication method, a communication device, a communication system, and a computer-readable medium. The method comprises: before a voice call establishment process or at an initial stage of the voice call establishment process, detecting whether a current signal meets communication quality requirements required for playing back a video customized alerting tone; and if the communication quality requirements are not met, sending, to a network device in a timely manner, a request message indicating that playing back the video customized alerting tone or a video customized ringing signal is not supported, so as to cancel the use of a customized alerting tone service or a customized ringing signal service during the call establishment. In this way, transmitting more messages or signaling between a user equipment and the network device for establishing QCI2 can be avoided, and reducing the transmission of related message signaling or media resources can reduce message or signaling overhead, thereby facilitating improving the call establishment success rate, improving the stability of voice calls on user equipment, and facilitating improving call experience of users.

The embodiment of the present application is suitable for the field of communication technology, and provides a call establishment method and an electronic device. The method is applied to the process of establishing a voice service between a terminal device and a first network device, and includes: in the case that the signal strength of a first detected signal between the terminal device and the first network device is less than a first threshold value, if the signal strength of the first detected signal is greater than a second threshold value, suppressing the reporting of a measurement report of a preset prediction event to the first network device, and if the signal strength of the first detected signal is less than or equal to the second threshold value, reporting the measurement report of the preset prediction event to the first network device; wherein the second threshold value is less than the first threshold value, the first threshold value is a threshold value for triggering the terminal device to report the preset prediction event to the first network device, and the preset prediction event is used for the first network device to instruct the terminal device to switch from the first network device to a second network device, so as to ensure the continuity of the voice service.

The invention discloses a communication method and terminal equipment, and relates to the field of communication. The method comprises the following steps: when a called terminal receives a call termination message from a network device, wherein the call termination message comprises that network resource bearing cannot be used for a video color vibration function or cannot support the video color vibration function, and a network area where the called terminal resides is a non-registration area, the called terminal closes the video color vibration function. After the called terminal closes the video color vibration function, if the called terminal receives another call request message from the network equipment, the called terminal does not need to establish a network resource bearer for the video color vibration function with the network equipment; therefore, the call establishment failure caused by the fact that the network resource bearer cannot be used for the video color vibration function or cannot support the video color vibration function is avoided, and the success rate of call establishment is improved.

The invention relates to a fire-fighting telephone extension interconnection system and method based on voice recognition. The system comprises an IP fire-fighting telephone switchboard, a plurality of fire-fighting telephone extensions, a voice recognition control server and a system database. The IP fire-fighting telephone switchboard is provided with a routing rule and is used for carrying out communication between the fire-fighting telephone extension set for primary routing call establishment and the voice recognition control server according to the routing rule and carrying out communication between the fire-fighting telephone extension set for secondary routing call establishment and the target fire-fighting telephone extension set; the voice recognition control server is used for carrying out man-machine voice interaction and recognition according to a set voice guide process and generating a control instruction; capturing a user intention by using a voice recognition technology; and the IP telephone technology is utilized to reconstruct the call routing process, so that revolutionary crossing from manual switching to intelligent direct dialing is realized on the basis of completely not changing the hardware of the traditional fire-fighting extension set, and a long-standing pain point in the field of fire-fighting emergency communication is solved.

The invention discloses a call management method and system for a public and private dual-mode terminal, and the method comprises the steps: initializing a public and private call concurrency control system service, and receiving a callback object injected by a public network and private network call service through an initialization module; when a call request exists, calling an interface of the public and special concurrent strategy management module by a call service to initiate permission query; the module combines current call information and priority information provided by the private network service priority conversion module and judges whether call establishment is allowed or not according to a preset strategy, and a high-priority new call can hang up the current call; the post-talk service is allowed to complete call setup and add information to service maintenance. The system comprises a public network call service, a private network call service and a concurrency control system service comprising three core modules. According to the application, the system coupling and maintenance cost are reduced, the real-time performance of the high-priority service is ensured, multiple dual-mode dual-standby scenes are adapted, and the expansibility and compatibility are improved.