Abstract: A mechanism is disclosed to adapt in real-time the encoding bitrate of a video source in an adaptive video streaming system with the aim of avoiding playback interruptions and to guarantee the best possible video quality. The mechanism is made of a feedback loop in which a controller computes the encoding bitrate that the sender (the video server) transmits to a receiver (client) employing a packet switching network such as the Internet. The mechanism can be used in video streaming systems employing scalable encoding (f.i. H.264 SVC, VP8, VP9) or multi bit-rate encoding (stream-switching). The automatic control that computes the video encoding bitrate can be executed either at the server or at the client.

Abstract: A mechanism is disclosed to adapt in real-time the encoding bitrate of a video source in an adaptive video streaming system with the aim of avoiding playback interruptions and to guarantee the best possible video quality. The mechanism is made of a feedback loop in which a controller computes the encoding bitrate that the sender (the video server) transmits to a receiver (client) employing a packet switching network such as the Internet. The mechanism can be used in video streaming systems employing scalable encoding (f.i. H.264 SVC, VP8, VP9) or multi bit-rate encoding (stream-switching). The automatic control that computes the video encoding bitrate can be executed either at the server or at the client.

Abstract: An actuator is disclosed which is required to implement, without errors, a generic rate-based packet sending algorithm over a packet switching network such as the Internet. Typical applications are audio/video streaming over UDP or TCP, audio/video conference over IP, Voice over IP, real-time data delivery, IP Television, Digital Video Broadcast over IP, client-server or peer-to-peer content distribution, Content Delivery Networks, Hybrid peer-to-peer/CDNs. Rate-based packet sending is of key importance for providing Quality of Service/Experience (QoS/QoE) over the Internet because it provides reduced queuing delays and jitters, and reduced buffer sizes both at application and network layers.

Abstract: An actuator is disclosed which is required to implement, without errors, a generic rate-based packet sending algorithm over a packet switching network such as the Internet. Typical applications are audio/video streaming over UDP or TCP, audio/video conference over IP, Voice over IP, real-time data delivery, IP Television, Digital Video Broadcast over IP, client-server or peer-to-peer content distribution, Content Delivery Networks, Hybrid peer-to-peer/CDNs. Rate-based packet sending is of key importance for providing Quality of Service/Experience (QoS/QoE) over the Internet because it provides reduced queuing delays and jitters, and reduced buffer sizes both at application and network layers.

Abstract: The invention herein described consists of a method to compute in a end-to-end fashion the sending rate of data, audio and video flows over a packet switching network such as the Internet Protocol Network. The sending rate is computed as a function of the data sent in the recent past and as a function of a probing function w(t) that is dynamically set by taking into account signals of network congestion and the network changing available bandwidth.

Abstract: A method and apparatus for congestion control of a Transmission Control Protocol (TCP) connection. A TCP source monitors received acknowledgments (ACKs) and uses header information associated with the ACKs to determine an amount of data delivered at a TCP receiver. The amount of delivered data, divided by the time requited to deliver the data, gives a measure of instant data rate. The instant data rate is input to a filtering and averaging process to generate an estimate of the end-to-end bandwidth share or data rate of the connection. For the entire duration of the connection, the TCP source updates its estimate of the end-to-end bandwidth share every time an ACK is received. The estimate of the end-to-end bandwidth share is used the reset a congestion window and a slow start threshold whenever packet loss is indicated such as whenever a timeout expires or whenever three (3) duplicate ACKs are received.

Abstract: The invention herein described consists of a method to compute in a end-to-end fashion the sending rate of data, audio and video flows over a packet switching network such as the Internet Protocol Network. The sending rate is computed as a function of the data sent in the recent past and as a function of a probing function w(t) that is dynamically set by taking into account signals of network congestion and the network changing available bandwidth.

Abstract: The invention herein described consists of an algorithm, which performs an end-to-end estimation of the bandwidth available in an end-to-end connection established between a server and a client via a packet switching network such as the Internet Protocol Network (IP). This algorithm is used to properly regulate the input rate at the send side. Typical applications are delivering best effort traffic over TCP, or audio and video traffic over RTP/UDP. The invention is particularly effective over wireless Internet and can be used in a content delivery system to dynamically choose the best server between a set of servers to satisfy the request of a client, or to select the best route in a content deliver or global hosting system.

Abstract: A method and apparatus for congestion control of a Transmission Control Protocol (TCP) connection. A TCP source monitors received acknowledgments (ACKs) and uses header information associated with the ACKs to determine an amount of data delivered at a TCP receiver. The amount of delivered data, divided by the time requited to deliver the data, gives a measure of instant data rate. The instant data rate is input to a filtering and averaging process to generate an estimate of the end-to-end bandwidth share or data rate of the connection. For the entire duration of the connection, the TCP source updates its estimate of the end-to-end bandwidth share every time an ACK is received. The estimate of the end-to-end bandwidth share is used the reset a congestion window and a slow start threshold whenever packet loss is indicated such as whenever a timeout expires or whenever three (3) duplicate ACKs are received.

Abstract: The invention herein described consists of an algorithm, which performs an end-to-end estimation of the bandwidth available in an end-to-end connection established between a server and a client via a packet switching network such as the Internet Protocol Network (IP). This algorithm is used to properly regulate the input rate at the send side. Typical applications are delivering best effort traffic over TCP, or audio and video traffic over RTP/UDP. The invention is particularly effective over wireless Internet and can be used in a content delivery system to dynamically choose the best server between a set of servers to satisfy the request of a client, or to select the best route in a content deliver or global hosting system.