摘要:
The system gets a request to add a communication node to an existing peer-to-peer communication session between two communication nodes. The communication node is added to the existing peer-to-peer communication session. The communication nodes in the existing peer-to-peer communication have not previously defined a mixing node. The communication nodes in the existing peer-to-peer communication session exchange a mixing score. A mixing node is determined based on the exchanged mixing score. Additional systems are defined which determine a mixing node when two peer-to-peer communication sessions are joined into a combined peer-to-peer communication session.
摘要:
A call processing system includes a call processing server. The call processing server processes calls for an internal network that employs SIP features and functions. The call processing server can receive calls from or send calls to one or more external communication endpoints that are not part of the internal network. However, the call processing server can associate a floating user agent with the communication from the external communication endpoint and lock the floating user agent to a gateway. After locking onto a gateway and initiating the call, the floating user agent can then publish call event status and receive SIP primitives similar to other SIP-enabled devices.
摘要:
One or more participants in a communication are authenticated using an authentication metric such as a face print or voice print. A single telecommunication address (or one for each participant) that is not associated with a communication device and is associated with at least one of the participants is determined. The telecommunication address (or addresses) is sent during the initiation of a communication session.Other embodiments provide for sending a single associated telecommunications address, individual identifiers of each of the participants, and a communication device name to better identify exactly who is calling. The system can also detect when an additional participant has joined the communication session and when a participant leaves the communication session.
摘要:
A call processing system includes a call processing server. The call processing server processes calls for an internal network that employs SIP features and functions. The call processing server can receive calls from or send calls to one or more external communication endpoints that are not part of the internal network. However, the call processing server can associate a floating user agent with the communication from the external communication endpoint and lock the floating user agent to a gateway. After locking onto a gateway and initiating the call, the floating user agent can then publish call event status and receive SIP primitives similar to other SIP-enabled devices.
摘要:
A call processing system includes a call processing server. The call processing server processes calls for an internal network that employs SIP features and functions. The call processing server can receive calls from or send calls to one or more external communication endpoints that are not part of the internal network. However, the call processing server can associate a floating user agent with the communication from the external communication endpoint and lock the floating user agent to a gateway. After locking onto a gateway and initiating the call, the floating user agent can then publish call event status and receive SIP primitives similar to other SIP-enabled devices.
摘要:
A call processing system includes a call processing server. The call processing server processes calls for an internal network that employs SIP features and functions. The call processing server can receive calls from or send calls to one or more external communication endpoints that are not part of the internal network. However, the call processing server can associate a floating user agent with the communication from the external communication endpoint and lock the floating user agent to a gateway. After locking onto a gateway and initiating the call, the floating user agent can then publish call event status and receive SIP primitives similar to other SIP-enabled devices.
摘要:
A method is presented for the automatic selection of the active software environment of a telecommunications terminal. In accordance with one embodiment of the present invention, the active software environment of a telecommunications terminal is selected on the basis of a characteristic of an incoming invitation to participate in a telecommunications session. In accordance with another embodiment of the present invention, the content of files residing in storage used by a system software instance is processed. When an incoming call is received, the present invention selects an active software environment on the basis of whether the caller is identified in any of the processed files.
摘要:
Methods and systems provided herein allow a communication session to be established on an optimal communication network and subsequently transferred to alternative networks if conditions dictate that such a change is necessary or desirable. The types of conditions which may be considered when identifying the optimal communication network or the alternative network include conditions related to the operation of a user device, communication profiles of the user, resources available on other networks, and the like.
摘要:
Voice analysis of a subscribers' greeting is used to assist with determining a true identity of a caller. When a greeting is recorded by the subscriber (e.g., the subscriber speaks their voice as part of their default greeting, or a custom greeting for a voice mail system), the system can analyze the greeting and create a voice signature or voiceprint of the greeting. This voiceprint information can be saved in the system and associated with the subscriber. When a subscriber changes their greeting that was previously analyzed to create a voiceprint, the messaging system can optionally analyze the newly recorded greeting to create a new voiceprint for the subscriber, with the system saving the new voiceprint in the system for future recognition tasks. This voiceprint is then used to identify the true identity of a caller that leaves a voice message.
摘要:
A communication manager establishes a call between two or more participants on two or more communication devices. The call can be an audio or video call. A call request is sent from one of the communication devices to conference an additional communication device to the call. For example, a participant in the call decides to conference an additional person to further discuss an idea.An information manager gets a roster of the participants. In addition to the call request, the roster of the participants is sent to the additional communication device. The additional communication device receives the call request and the roster. The roster is displayed to the user of the additional communication device. The user can then indicate to answer the call request. If answered, the additional communication device sends an indication that the call was answered and the additional communication device is conferenced into the call.