Abstract: The present invention relates to systems and methods for improving transmission of voice packets over a network are provided. The systems and methods include a few central internet data centers (IDCs) which include a routing controller and an access controller. The system also includes a number of edge IDCs. Each edge IDC includes a last mile optimizer and a relay server. The last mile optimizer operates along with an application located on the users' devices and the access controller in the central IDCs to identify the best edge server for the particular device to connect to. The edge servers continually monitor pathway performance once the call is in progress. If an error is detected, then the server may automatically transition to back-up pathways rapidly to minimize call performance disruption.

Abstract: System and methods of managing reference frame adaptively for video communication under a network are provided. The system adaptively decides the interval between each I frame, the bitrate of the encoded bit steams, and the resolution of the encoded video to achieve quick connection. The system may set the appropriate reference frame(s) for the current frame, manage the reference frame(s) in the DPB to be long/short term reference frame(s) or mark them as unused for reference; mark the current frame to be a long term reference frame, a short term reference frame, unused for reference or skip encoding the current frame; adaptively change the ratio of long term reference frame(s) with respect to the short term reference frame(s) in the DPB; adaptively change the frequency ratio of marking the current frame as long term reference frame and marking the current frame as short term reference frame.

Abstract: A method for encoding multiple descriptions for a media stream includes: determining, for a current block of the media stream, a first number of duplicate blocks, determining, for the current block, a plurality of complementary forward gain factors, and determining a first number of descriptions for the current block by applying the plurality of complementary forward gain factors to the first number of duplicate blocks. A method for decoding multiple descriptions for a media stream includes: determining, for a current block of the media stream, a first number of descriptions from the media stream, in which each description comprises a scaled block corresponding to the current block, determining backward gain factors for the first number of scaled blocks, and determining the current block by applying the backward gain factors to the first number of scaled blocks.

Abstract: A decoding method includes: decoding from the video bitstream a prediction status parameter indicative of whether SOP is enabled; if the SOP is enabled, determining a SOP type indicative of a specific operation for predicting a current picture of the video bitstream; and determining a predicted picture by applying the specific operation to at least one of a reference picture for the current picture and the current picture, in which the specific operation includes at least one of rotation and zoom. An encoding method includes: if SOP is enabled, determining SOP data indicative of a specific operation for predicting a current picture of a video sequence; determining a predicted picture by applying the specific operation to at least one of a reference picture for the current picture and the current picture, in which the specific operation includes at least one of rotation and zoom; and encoding the SOP data.

Abstract: A method for decoding multiple descriptions for a media stream includes decoding at least one of a first description and a second description from the media bitstream, wherein the first description and the second description are associated with a current sample of the media bitstream and independently decodable; based on a determination that the second description is missing, reconstructing, by a decoder, the current sample by performing a third shifting operation on the first description, wherein the third shifting operation on the first description comprises left shifting the first description by one bit; and based on a determination that both the first description and the second description have been received, reconstructing, by the decoder, the current sample by summing the first description and the second description. An encoding method is also provided.

Abstract: Systems and methods for improving quality of a call over network (CON) are provided. Call quality may be improved via buffer length modulation based upon the call scenario type. Scenario detection may be based upon who speaks, and the duration of the speaking, as well as contextual analysis. Further, the call over network quality may further be improved by deploying modules over the network. The modules are intermediary vehicles between each communicator and backend servers. The modules intercept audio packets from the communicator to detect packet loss, and perform recovery of lost packets, thereby accelerating real-time audio conversations.

Abstract: A real-time communication monitoring and diagnosing system is provided on a real-time communication monitoring and diagnosing computer system. The monitoring and diagnosing system includes time line, a video downstream bitrate axis extending upward from reference line, an audio downstream bitrate axis extending downward from the reference line, a video downstream bitrate curve in a first color above the reference line, an audio downstream bitrate curve in a second color below the reference line, a set of choppy video marks in a third color above and along the reference line, a set of blurry video marks in a fourth color as a dotted line along the video downstream bitrate curve, and a set of audio marks in a fifth color below and along the reference line.

Abstract: A method for encoding a video stream, including encoding, by a processor, a first slice of a first picture of the video stream; receiving, from a decoder, a feedback message indicative of having received the first slice of the first picture; and in response to receiving the feedback message indicative of having received the first slice of the first picture, setting the first slice of the first picture as a reference slice for encoding a second picture of the video stream, wherein the second picture is encoded after receiving the feedback message.

Abstract: A method for decoding a compressed video stream is disclosed. The method includes decoding, by a processor from the compressed video stream, a first picture and a second picture, each including multiple slices in the same pattern, in which the first picture includes a first I-slice decoded using intra-prediction, the second picture includes a second I-slice decoded using intra-prediction, and the first I-slice and the second I-slice are not co-located with each other; and decoding, using inter-prediction, a first P-slice of the current picture referencing the first I-slice, and a second P-slice of the current picture referencing the second I-slice, in which the first P-slice of the current picture is co-located with the first I-slice, and the second P-slice of the current picture is co-located with the second I-slice.

Abstract: A method for encoding a video stream, including encoding, by a processor, a first slice of a first picture of the video stream; receiving, from a decoder, a feedback message indicative of having received the first slice of the first picture; and in response to receiving the feedback message indicative of having received the first slice of the first picture, updating a dynamic reference picture (DRR) with the first slice of the first picture, wherein the first slice of the first picture replaces the first slice of a second picture in the dynamic reference picture, and the first slice of the first picture and the first slice of the second picture are co-located.

Abstract: The present invention relates to systems and methods for network labeling in order to enhance real time data transfers. A network for a real time data transfer is identified and predictive models for network performance are compared against to determine if the network is suitable for the data transfer. If so, then the real time data transfer may be completed as expected. However, if the network is predicted to be unsuitable for transmission an alternate means for connection may be suggested. The alternate suggestion may include delaying the data transfer until the network is expected to be in better conditions, connecting to another access point in the network, or switching to another network entirely. During the data transfer, the quality of the network is monitored in order to update the predictive models for the network's quality. Identifiers for the network may be utilized to keep track of the networks.

Abstract: The present invention relates to systems and methods for best server selection for real-time data transmission are provided. The systems and methods include an access controller, and an application which receives information of a selection of several best servers with which to transmit real-time packets to. The application sends a request to the access controller for server information before call initiation. The access controller identifies servers with the same ISP, and then selects a regional server within this group of same ISP servers. Subsequently the geo optimizer identifies servers closest to the application from among the regional servers. The respective loads of each of the closest servers are analyzed to select only servers that are likely to provide satisfactory quality of service. A monitor within the central IDC is able to measure load conditions of each server and provide this load information to the access controller.

Abstract: The present invention relates to systems and methods for last mile optimization for the transmission of real-time data. Pseudo-packets are sent to a last mile optimizer located at each of several ‘best’ servers. The last mile optimizers provide feedback regarding jitter, latency and packet loss of the pseudo-packets to determine quality of service for each of the servers. The server with the best quality of service is selected as the ‘single best’ server. A plurality of transporter algorithms may then be applied to pseudo-packets that are then sent to the single best server, and the quality of service for each transporter algorithm is subsequently measured for effectiveness. The transporter algorithms that are shown to be effective are combined together for the actual transmission of real-time data. Continuous monitoring of transmission quality allows for utilizing backup pathways upon detection of a problem, and negotiation of preferred transporter algorithms.

Abstract: Systems and methods for a readiness dialog box for a call over network (CON) are provided. In some embodiments, the readiness dialog box is presented to the callers prior to the onset of the call. It presents the other participant's and their status. For example, it may indicate which participants are online, but not yet ready, those who are ready, and those not available. It also enables the caller to send messages (both preconfigured and customized) to the other participants. Once sufficient participants have joined, the call may start. Sufficiency of participants could include a quorum of individuals, may require that specific participants are ready, everyone is ready, or may be time dependent. The participant requirements may be configured by the individual setting up the call based upon call type.

Abstract: Systems and methods for improving quality of a call over network (CON) are provided. Call quality may be improved via buffer length modulation based upon the call scenario type. Scenario detection may be based upon who speaks, and the duration of the speaking, as well as contextual analysis. Further, the call over network quality may further be improved by deploying modules over the network. The modules are intermediary vehicles between each communicator and backend servers. The modules intercept audio packets from the communicator to detect packet loss, and perform recovery of lost packets, thereby accelerating real-time audio conversations.

Abstract: Systems and methods for improving quality of a call over network (CON) are provided. Modules may be distributed over the network at nodes. The modules monitor data quantity through the node, and data quality from one node to another. This fidelity and usage data is provided to a server where the data may be utilized to decide on how to route a call over network. The selected pathway may be decided based upon avoidance of “problem” legs where packets have historically been lost, or based upon the desire to load level data transmissions. A signal indicator for the signal quality of the networks “last mile” may be provided to the user. The last mile is the portion of the data pathway dependent upon the telephone service provider's infrastructure that cannot be rerouted around.

Abstract: Systems and methods for a call over network (CON) with a visualized summary are provided. In some embodiments, after the call concludes, a visualized summary of the call can be generated. The summary includes any of the recording of the call, transcriptions, scenario information, speaker information and the duration each speaker was talking, etc. Scenario may be generated by comparing the call speaking pattern to known templates of call types. Additionally, participant features employed in the call may be summarized in chronological relation to the calls progression in a visual format.

Abstract: Systems and methods for visualization of a call over network (CON) are provided. In some embodiments, the visualization of a call over network may be effectuated by three functionalities: a readiness dialog box that enables the facilitation of the call, the inclusion of visualization and participant features within the call, and a visualized summary after the call. The readiness dialog box is presented to the callers prior to the onset of the call. It presents the other participant's and their status. It also enables the caller to send messages (both preconfigured and customized) to the other participants. Once sufficient participants have joined, the call may start. Once the call starts, it may be visualized by displaying on a single or multi channels, which caller is speaking, and any additional indications they may be providing. The caller may likewise be provided a set of participant features that allow the user to interact with the call.

Abstract: Systems and methods for improving quality of a call over network (CON) are provided. Call quality may be improved via pathway testing to determine data path quality. This may be utilized to inform buffering lengths, and also may be utilized to choose the data pathway utilized for transmitting the data. Pathway testing may employ collecting microphone data on one device, transmitting it across the various pathways, and then comparing the quality at the endpoint compared to the initial data.