Abstract:
Anchoring a communication session for a target mobile phone includes identifying a current access node operable to provide the target mobile phone access to a first network of a first network type. Potential access nodes operable to provide a mobile phone access to a second network of a second network type are identified. Whether the communication session can be handed off to a potential access node of the one or more potential access nodes is established. Anchoring of the communication session is initiated in the second network only if the communication session can be handed off to a potential access node of the one or more potential access nodes.
Abstract:
In one embodiment, a method includes identifying a critical time when a current presence state associated with a first client is scheduled to change to a timed presence state. The method also includes generating a first presence document before the critical time, and providing the first presence document to at least a second client. The first presence document has an indication of the current presence state and the timed presence state, and is provided to the second client before the critical time.
Abstract:
A method and system for dynamically selecting a destination gateway to complete a call over a path supported at least in part by an IP telephony network and a public switched telephone network. The method and system further provide for dynamically detecting available gateways, dynamically removing failed and/or unavailable gateways, and automatically recovering failed and/or unavailable gateways after a predetermined period of time. A method is also provided for detecting available destination gateways using a ping method, where a message is transmitted to a plurality of destination gateways on a one-by-one basis to ascertain the availability status of each destination gateway.
Abstract:
A method and system for providing quality of service in an IP telephony session between a calling party and a called party establishes a high quality of service ATM virtual circuit for the session between first and second devices, each of the devices having ATM capability and IP capability. The first and second devices provide bidirectional translation between IP media and ATM media. The system transports IP media for the session between the calling party and the first device, and between said called party and a second device. The virtual circuit transports ATM media for the session between the first and second devices. An intelligent control layer provides IP and ATM signaling to set up the session.
Abstract:
A method and system for providing quality of service in an IP telephony session between a calling party and a called party establishes a high quality of service ATM virtual circuit for the session between first and second devices, each of the devices having ATM capability and IP capability. The first and second devices provide bidirectional translation between IP media and ATM media. The system transports IP media for the session between the calling party and the first device, and between said called party and a second device. The virtual circuit transports ATM media for the session between the first and second devices. An intelligent control layer provides IP and ATM signaling to set up the session.
Abstract:
In one embodiment, a method includes receiving a message associated with a device in a first domain. An identifier is determined for the device. The message is then sent to a load balancer where the message includes the identifier. The load balancer is then configured to send the message to a network device in a plurality of network devices. The network device is configured to process messages from the device. A second message may be received at the load balancer from a second domain. The second message may include the identifier for the device. The load balancer may then send the second message to the selected network device such that the first message and the second message are processed by the same network device.
Abstract:
The present invention discloses a method whereby the separate protocols: session initiation protocol SIP, resource reservation protocol RSVP, common open policy service COPS, and open settlement protocol OSP are used together to setup, maintain, and teardown Internet communications having an acceptable QoS. This process is accomplished by dynamically establishing RSVP policy based on SIP telephony requests to provide IP communications with QoS across the Internet. The QoS policy is installed in network elements at the request of the network elements. The network elements receive a RSVP PATH or RESV request and queries the policy server; the policy server queries a Local database about ID and services for the user and a clearinghouse server (if available) or a policy server in a corresponding network; upon positive acknowledgement from the local database and/or the clearinghouse server, the policy server confirms policy in network elements to accept RSVP PATH and RESV requests for the particular reserved data flow to the SIP client. In this manner, the called telephone will not ring until policy has been provisioned in the network elements and resources have been reserved end-to-end to ensure an acceptable level of QoS.
Abstract:
A method for signaling an Integrated Messaging System (IMS) on an Internet Protocol (IP) based network to deposit a message, including the steps of sending a Session Initiation Protocol (SIP) SIP INVITE request to the IMS indicating a message deposit action; receiving a corresponding SIP message from the IMS agreeing to participate in the message deposit action; and sending an SIP acknowledge message to the IMS confirming receipt of the corresponding SIP message; and depositing the message in a destination mailbox. A method of signaling an IMS on an IP based network to retrieve a deposited message, the method including the steps of sending a SIP INVITE request to the IMS indicating a message retrieval action; receiving a corresponding SIP message from the IMS agreeing to participate in the message retrieval action; sending an SIP acknowledge message to the IMS confirming receipt of the corresponding SIP message; and retrieving the deposited message from a mailbox corresponding to known account information.
Abstract:
An illustrative intelligent network and method for providing voice telephony over ATM and closed user groups are provided that can provide significant advantages. A method for providing a closed user group service to authorize VToA calls includes determining the closed user group identifiers for a calling party, determining the closed user group identifiers for a called party, locating a common closed user group identifier that is common to the calling party and the called party, analyzing the privileges of the calling party in the common closed user group to determine if the calling party can make calls to other users of the common closed user group, and analyzing the privileges of the called party in the common closed user group to determine if the called party can receive calls from other users of the common closed user group. An illustrative intelligent network and data structure to provide closed user group services is also provided.
Abstract:
A method for combining Internet protocols in a Differentiated Services model environment is described. The Session Initiation Protocol (SIP) and Common Open Policy Service (COPS) are combined together to provide methods of setting up a session and tearing down a session, while maintaining Authentication, Authorization, and Accounting (AAA) policies. The Open Settlement Policy (OSP) is also combined with SIP and COPS. This combination provides for an interchange of parameters between session setup, teardown, authorization, policy, Quality of Service (QoS), and usage reporting