Cisco Technology, Inc.

Ali C Begen

David R Oran

William C Ver Steeg

Joshua B Gahm

Tankut Akgul

Michael Anthony Ramalho

Arturo A Rodriguez

Daniel George Wing

Scott C Labrozzi

Eric Cunningham

Srinath Gundavelli

Gregory J Shepherd

Toerless Eckert

Zhi Li

Francois Le Faucheur

Biswaranjan Panda

Frederick Juergens Baker

Mark John Baugher

Bruce P Osler

Wilson Chung

Bill Ver Steeg

Saamer Akhshabi

In one embodiment, a method that receives a manifest for plural encoded representations of a single content stream, each representation fragmented into plural chunks, each representation comprising a different quality level, the manifest listing a plurality of representations, each representation comprising the plural chunks at one of a plurality of quality levels, and requests one of the plural chunks based on selection of one of the plurality of quality levels explicitly indicated in the manifest.

Streaming manifest quality control

In one embodiment, an encapsulator device receives a source stream of encoded packets in a first transport format and converts the source stream to packets of an output stream in a second transport format for communication over a data network. The encapsulator device generates a mapping stream with information indicating which packets in the source stream are used to generate the output stream. The output stream of the encapsulator device is coordinated with the output stream of other encapsulator devices by sending the mapping stream to the other encapsulator devices in the network. The other encapsulator devices use the mapping stream to generate identical output streams.

Distributed coordination of network elements for packet encapsulation

A method is provided in one example embodiment and includes receiving at a first network node a request to obtain data from a second network node, wherein the first and second network nodes are connected via n access networks; partitioning the request into n subrequests proportionally based on relative throughputs of the n access networks; and transmitting each of the n subrequests to the second network node via a respective one of the n access networks.

Hypertext transfer protocol support over hybrid access

In one embodiment, a client device includes a processor to send a first GET request to a server for a first data chunk of a data item, a communication interface to receive packets of the first data chunk, a buffer to buffer the packets, wherein the processor is operative to obtain a first measure of round trip time (RTT), the processor is operative to estimate a time twhen receipt of the last packet of the packets of the first data chunk in the buffer will be completed, the processor is operative to determine a next fetch time, t, to send a second GET request to the server for a second data chunk of the data item, based on the time tadjusted by a portion of the first measure of RTT, the processor is operative to send the second GET request to the server at about time t

Pipelining get requests in adaptive streaming

In one method embodiment, receiving a video stream comprising a first compressed picture without associated time stamp information and a second compressed picture having associated first time stamp information, the second compressed picture following the first compressed picture in transmission order; deriving second time stamp information based on the first time stamp information; and processing the first compressed picture based on the second time stamp information.

DTS/PTS backward extrapolation for stream transition events

A method is provided in one example embodiment and includes communicating content to a content receiver, where a first portion of the content has a first drop priority. The method also includes determining a buffer status for a buffer at the content receiver, and adjusting the first drop priority to a second drop priority for a second portion of the content based on the buffer status. In more particular embodiments, the second drop priority can be higher than the first drop priority of the first portion of the content if the buffer is full or above a threshold. Also, the second drop priority can be lower than the first drop priority of the first portion of the content if content in the buffer has not started to render.

System and method to reduce stream start-up delay for adaptive streaming

In one method embodiment, providing a multiplex of compressed versions of a first video stream and a first audio stream, each corresponding to an audiovisual (A/V) program, the first video stream and the first audio stream each corresponding to a first playout rate and un-synchronized with each other for an initial playout portion; and providing a compressed version of a second audio stream, the second audio stream corresponding to a pitch-preserving, second playout rate different than the first playout rate, the second audio stream synchronized to the initial playout portion of the first video stream when the first video stream is played out at the second playout rate, the first audio stream replaceable by the second audio stream for the initial playout portion.

Preserving synchronized playout of auxiliary audio transmission

A method is provided in one example embodiment and includes receiving video data at an adaptive bitrate (ABR) client that includes a buffer; determining whether a buffer level for the buffer is below a target buffer level; applying a random delay for a fetch interval associated with requesting the video data; and requesting a next segment of the video data after the random delay. The random delay can provide for a plurality of fetch times to become decorrelated from each other.

System and method for providing randomization in adaptive bitrate streaming environments

Techniques are provided for optimizing quality in adaptive streaming, where information is obtained relating to encoding bitrates and content quality scores for content segments of a content stream. The content segments of the content stream are provided by a content server and downloaded by a computing device. For each content segment, an encoding bitrate is determined for downloading the content segment from the content server, where determining the encoding bitrate for a current content segment is based upon an estimated current network bandwidth, constraints of a buffer into which downloaded content segments are stored, and also upon quality scores associated with content segments within a selected horizon extending from the current content segment to a future content segment in the content stream at an endpoint of the horizon. Each content segment with the determined bitrate for the content segment is downloaded from the content server.

Quality optimization with buffer and horizon constraints in adaptive streaming

A method is provided in one example embodiment and includes receiving media from an input; gathering metadata from a plurality of media stream representations that are provided by the input; creating one or more metadata segments corresponding to the plurality of media stream representations in order to generate a metadata track; and making at least a portion of the metadata track available to a client device, where the portion of the metadata track is provided separate from a manifest file.

Use of metadata for aiding adaptive streaming clients

A method is provided in one example and includes receiving a loss report usable to determine a plurality of missing packets at a first receiver; determining one or more retransmission packets for retransmission based on a forward error correction (FEC) scheme and the plurality of missing packets at the first receiver, where the one or more retransmission packets are determined to allow the first receiver to recover at least one of the missing source packets in the one or more source packets using the FEC scheme; and sending the one or more retransmission packets to the first receiver, the one or more retransmission packets allowing the first receiver to recover at least one of the missing source packets in the one or more source packets that it is missing using the forward error correction scheme after receiving the one or more retransmission packets.

Importance-based FEC-aware error-repair scheduling

In one embodiment, a method includes identifying a current encoding rate requested by a client device for content received from a content source, setting at a network device a rate limit to limit the rate at which the content is received at the client device based on the current encoding rate, and adjusting the rate limit based on changes in the current encoding rate. The rate limit is set to allow the client device to change the current encoding rate to a next higher available encoding rate.

Stabilization of adaptive streaming video clients through rate limiting

In one embodiment, a method includes transmitting media from a source to a plurality of receivers in a first source-specific multicast (SSM) channel and transmitting from the source to said plurality of receivers, a channel change message in an in-band transport control protocol message identifying a second SSM channel, wherein the receivers are configured to join said second SSM channel in response to said channel change message.

Method and apparatus for in-band channel change for multicast data

Content is received within a network at a content receiver and that is communicated from a content source, where the content comprises packets to be sent to the content receiver that are marked with a first drop priority or a second drop priority. A network priority is detected at the content receiver that is based at least in part upon a ratio of packets marked with the first drop priority to packets marked with the second drop priority within the content and also a determination at the content receiver of a rate of packets dropped that are marked with the first drop priority and a rate of packets dropped that are marked with the second drop priority.

Growing community of inventors

London, Canada

Ali C Begen