Abstract: An architecture and method is provided for call routing using both IMS and non-IMS frameworks. The method includes receiving presence information of a third party from a non-IP Multimedia Subsystem (IMS) network device. The method further includes routing the third party to at least one callee designated device based on configurable preferences provided by the callee and correlated to presence information using an IMS compliant component. The method additionally includes providing a charging record for the routing on an IMS complaint charging platform.

Abstract: The present invention provides for routing calls between disparate domains, such as a circuit-switched subsystem and a multimedia subsystem. When a user element is homed in a first domain and roaming in a second domain, an incoming call will arrive at a gateway node in the first domain. As a result, a message identifying the user element and indicating that an incoming call has been received at first gateway node for the first domain is sent to a continuity control function (CCF) residing in the multimedia subsystem. The CCF will create and effect delivery of an inter-domain routing number to the first gateway node. The inter-domain routing number is used by the first gateway node to route the call to the second domain. The inter-domain routing number may be associated with a second gateway node of the second domain.

Abstract: A method for establishing a call over a packet network may include receiving a call request via an originating trunk on a packet network from an originating call device. Status of an originating trunk and terminating segment may be determined. If the status of the originating trunk and terminating segment are within a first range, the call between the originating and terminating call device over the determined transmission path at a first data rate may be established. Otherwise, if the status of the originating trunk or terminating segment is within a second range, a determination as to whether the originating trunk and terminating segment can operate at a second, lower data rate may be made, and, if so, the call may be established over the transmission path between the originating call device and terminating call device at the second, lower data rate.

Abstract: A packet switch collects configuration information on the peer devices to which it connects. Peer devices that are IP (Internet Protocol) telephones are sensed. Such devices may also bridge packets to a connected device such as a computer, but with a bridge capability that cannot handle large broadcast storms on the network. The packet switch uses the configuration information to limit broadcast storms on its ports connected to the relatively fragile IP telephones/bridges. This can prevent broadcast storms from disrupting calls on the IP telephones connected to the packet switch.

Abstract: There is described a transmission of user data from a source communications device provided with a first encoder for encoding users data to a target communications device provided with a first decoder for decoding said user data via a communication network which is provided with several converting devices comprising additional encoders and additional decoders for carrying out a verification of the converting devices. Via the verification it is determined, whether the first encoder is compatible with the decoder of a given converting device and, whether the first decoder is compatible with the encoder of said converting device. One of the converting devices for which the compatibility is ascertained by the verification is selected for transmitting user data.

Abstract: A system and method for recording analog signals exchanged between a telephone device and a VoIP device, capturing packets exchanged between the VoIP device and an IP network, determining analog time values corresponding to analog characteristics of the analog signals, determining digital time values corresponding to digital characteristics of the packets, determining a common reference time for the analog time values and digital time values and determining a processing delay based on the analog time values and the digital time values.

Abstract: An indirect communication system and a method of indirect communication include a mobile phone as either a calling device or receiving device. The mobile phone calls another device to set up a prospective communication, and then uses Voice over Internet Protocol (VoIP) to communicate with the other device over the Internet. The receiving device receives a generated signal notifying the receiving device of a proposed communication with the calling device. A server sets up a meeting point channel after the calling device has connected to the server. The server receives outgoing VoIP packets from the calling device and redirects the outgoing VoIP packets to the receiving device.

Abstract: A unified session detail records of a multi-hop session is provided. The multi-hop session may be established through multiple nodes in a VoIP network. Each node may generate session detail records for the legs of the multi-hop session that are connected to the node. The nodes on the path of the multi-hop call may send the record to a single node to consolidate the session detail records of the legs of the multi-hop call in the single node so that the single node can provide unified session detail records of the multi-hop session. By providing unified session detail records, the user does not need to track down the nodes that are on the path of the multi-hop call to obtain the session detail records of the multi-hop session.

Abstract: Disclosed are conferencing systems, devices, architectures, and methods for managing a conference spanning system to facilitate participation in a plurality of conferences at the same time. An exemplary system having devices and/or structures in accordance with embodiments can include: (i) a plurality of conferences, each conference being configured to support a plurality of media modes; (ii) a presentation mechanism configured to support the plurality of media modes from the plurality of conferences; and (iii) a selector for selecting a conference from the plurality of conferences and designating the selected conference as an active conference and designating non-selected conferences as background conferences, wherein the active conference is assigned at least one enhanced media mode in the presentation mechanism.

Abstract: A method and apparatus for performing a active packet bundling in a VoIP (Voice over Internet Protocol) based on a measure of voice concealability of one or more speech packets to be transmitted. Voice concealability is a property of a speech packet which indicates the likelihood that a given speech segment will meet the “quasi-stationary” (QS) behavior assumed by conventional Packet Loss Concealment (PLC) algorithms used to compensate for transmission loss of speech packets. By ensuring that only well-concealed frames are delayed for bundling, it is more likely that the receiver will run the PLC algorithm when it is most effective (i.e., when it is likely to perform well at reconstructing the missing speech frame), rather than when it is more likely to be ineffective. This allows for a reduction in bandwidth while minimizing any potential negative impact on quality.

Abstract: A system and method for routing a call over a packet network. A call request may be received from a calling party to call a called party at a network address. At least one potential call path over a packet network may be determined to connect the calling party to the called party at the network address. Network performance information associated with each potential call path may be accessed and a determination may be made that each of the call paths are impaired or congested. In response to determining that each of the call paths are impaired or congested, the call may be routed over a call path other than one of the at least one potential call paths to enable the calling party to communicate with the called party.

Abstract: A method of monitoring quality of a telephone call transmitted over a network includes obtaining a metric associated with the telephone call, where the metric is based on data packets that are transmitted during the telephone call, and determining whether the metric exceeds a threshold. Plural metrics may be obtained and the method may determine whether the plural metrics exceed corresponding thresholds.

Abstract: A system (10), method (90) and apparatus (14) are provided for provisioning a multicast stream, such as an IP multicast stream, in a network. The system includes a first tier (12), such as a national backbone, with a first Multicast Address Discovery (MCAD) directory (34), and at least one second tier having a headend, such as a regional headend (14), with a second MCAD directory (36, 38). The system (10) includes a tiered MCAD directory network that allows the assignment of multicast addresses to multicast streams received from an upper tier MCAD directory. The tiered MCAD directory structure allows for requesting and storing stream descriptors and delivering stream descriptors on-demand to MCAD-enabled devices throughout the system. The system (10) also allows for stream provisioning maintenance and validation at lower tier headends via MCAD-enabled device queries or re-queries of upper tier MCAD directories.

Abstract: Systems and methods of providing link adaptation information feedback are provided. A mobile device that receives packets generates link adaptation information based on incorrectly received packets. This can involve sending link adaptation information in association with NACKs (negative acknowledgements) generated by the mobile device. The network receives this link adaptation information and performs link adaptation accordingly.

Abstract: A method and system for updating physical location data associated with a VoIP endpoint device is implemented in a variety of embodiments. In one such embodiment, a VoIP endpoint device stores an identifier for a packet-communicating device which is subject to changing. The VoIP endpoint device has an interface for communication over the Internet via the packet-communicating device and uses a circuit-implemented method for prompting a VoIP user to update physical location data associated with the VoIP endpoint device. In response to a power state transition and a change in the stored identifier, the VoIP endpoint device facilitates an update to the physical location data associated with the VoIP service.

Abstract: A method of ringcasting with improved reliability, in which a participant who is the source of content saves the signal in which the content is found, and compares received signals with the signal in which the content is found to ensure that the content has made it completely around the ring. If any particular piece of the content does not complete the ring, the content is re-transmitted from the participant around the ring. Sequence indices within the signal informs each participant of the sequence in which any particular piece of content is to be received so that, even if any particular piece of content is received out of order, it will be replaced in its proper sequence by a recipient.

Abstract: A method of transmitting a combined audio signal to at least one of a plurality of participants in a communication event comprising; receiving an audio signal from each of said plurality of participants together with audio activity information associated with each of said received audio signals; analysing a measure of audio activity for each received audio signal based on the audio activity information associated with each of said received signals, wherein said measure of audio activity allows audio signals comprising audio activity to be compared; selecting a set of audio signals from said received audio signals based on the analysed measure of audio activity for each signal; decoding said set of audio signals; and combining said set of audio signals to generate said combined audio signal to be transmitted to said at least one of said plurality of participants.

Abstract: In a particular embodiment, interworking between an H.323 protocol and SIP protocol for a call forward/redirection is provided. A first user may call a second user. The second user may have set preferences that cause a call redirection/forward to another end device. The end device may be associated with the second user or another user. Different preferences for the call forward/redirection may be provided. For example, a call may be forwarded all the time, when the second end device is busy, when there is no answer, etc. The redirection may then be performed using a gateway. When end devices communicate using SIP and H.323, the gateway provides interworking to provide the call forward/redirection. For example, the gateway maps a 3xx message in SIP to an H.450.3 message in the H.323 protocol, and vice versa, to provide the call forward/redirection.

Abstract: Various embodiments of present invention relate to the synchronization of an SDP stream to the media stream. In various embodiments, the SDP-consuming application initiates a FLUTE receiver and retrieves new versions of the SDP file. The terminal also retrieves information from the RTCP layer about the current sender time. The sender time is indicated as an NTP timestamp in RTCP sender reports. The receiver indicates to the player that the new SDP file will be valid after some time period. Various embodiments of the present invention also cover the behavior of a terminal upon receiving an SDP update. Still further, various embodiments of the present invention provides for richer out-of-band ESG delivery than conventional systems, permitting more meaningful session updates.

Abstract: The present invention provides a technique where application servers can initiate sessions within the IMS network by initially sending session initiation messages to the I-CSCF. By routing the session initiation messages to the I-CSCF, the application servers need not directly access the HSSs in order to initiate sessions. Upon receiving a session initiation message from an application server, the I-CSCF may access the HSS to identify an S-CSCF to use for session control and then route the session initiation message to that S-CSCF. The S-CSCF may then access the same or different HSS to determine how to further route the session initiation message to establish the session. The session initiation message may be routed toward other S-CSCFs en route to the appropriate user elements. The session may be a session established between user elements or between the application server and a user element.