US20010023454A1 - Codec-independent technique for modulating band width in packet network - Google Patents
Codec-independent technique for modulating band width in packet network Download PDFInfo
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- US20010023454A1 US20010023454A1 US09/181,947 US18194798A US2001023454A1 US 20010023454 A1 US20010023454 A1 US 20010023454A1 US 18194798 A US18194798 A US 18194798A US 2001023454 A1 US2001023454 A1 US 2001023454A1
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
- H04L69/00—Network arrangements, protocols or services independent of the application payload and not provided for in the other groups of this subclass
- H04L69/04—Protocols for data compression, e.g. ROHC
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
- H04L1/00—Arrangements for detecting or preventing errors in the information received
- H04L1/0001—Systems modifying transmission characteristics according to link quality, e.g. power backoff
- H04L1/0006—Systems modifying transmission characteristics according to link quality, e.g. power backoff by adapting the transmission format
- H04L1/0007—Systems modifying transmission characteristics according to link quality, e.g. power backoff by adapting the transmission format by modifying the frame length
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
- H04L47/00—Traffic control in data switching networks
- H04L47/10—Flow control; Congestion control
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
- H04L47/00—Traffic control in data switching networks
- H04L47/10—Flow control; Congestion control
- H04L47/12—Avoiding congestion; Recovering from congestion
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
- H04L47/00—Traffic control in data switching networks
- H04L47/10—Flow control; Congestion control
- H04L47/24—Traffic characterised by specific attributes, e.g. priority or QoS
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
- H04L47/00—Traffic control in data switching networks
- H04L47/10—Flow control; Congestion control
- H04L47/36—Flow control; Congestion control by determining packet size, e.g. maximum transfer unit [MTU]
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
- H04L65/00—Network arrangements, protocols or services for supporting real-time applications in data packet communication
- H04L65/60—Network streaming of media packets
- H04L65/65—Network streaming protocols, e.g. real-time transport protocol [RTP] or real-time control protocol [RTCP]
-
- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
- H04L65/00—Network arrangements, protocols or services for supporting real-time applications in data packet communication
- H04L65/80—Responding to QoS
-
- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
- H04L65/00—Network arrangements, protocols or services for supporting real-time applications in data packet communication
- H04L65/1066—Session management
- H04L65/1101—Session protocols
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- Engineering & Computer Science (AREA)
- Computer Networks & Wireless Communication (AREA)
- Signal Processing (AREA)
- Multimedia (AREA)
- Quality & Reliability (AREA)
- Computer Security & Cryptography (AREA)
- Data Exchanges In Wide-Area Networks (AREA)
Abstract
The size of packet payloads are varied according to the amount of congestion in a packet network. More data is put in packet payloads when more congestion exits in the packet network. When network congestion is high, less network bandwidth is available for transmitting packets. Accordingly, the packet payloads are transmitted with larger payloads to reduce the percentage of overhead in each packet. When there is little or no network congestion smaller packet payloads are transmitted. The additional overhead created in transmitting smaller packets is acceptable when there is little or no network congestion because the network currently has excess bandwidth. Thus, the packet payloads are dynamically adjusted to use network resources more effectively.
Description
- This invention relates generally to packet networks and more particularly to a system for adapting packet payload size to the amount of network congestion.
- A data stream is transmitted over a packet network by first formatting the data stream into multiple discrete packets. For example, in Voice Over Internet Protocol (VOIP) applications, a digitized audio stream is quantized into packets that are placed onto a packet network and routed to a packet telephony receiver. The receiver converts the packets back into a continuous digital audio stream that resembles the input audio stream. Acodec (a compression/ decompression algorithm) is used to reduce the communication bandwidth required for transmitting the audio packets over the network.
- A large amount of network bandwidth is required for overhead when a data steam is converted and transmitted as packets. For example, in Realtime Transport Protocol (RTP)-encapsulated VoIP, a very common codec technique packetizes two 10 millisecond (ms) frames of speech into one audio packet. For a 8 kilobit per second (Kbit/s) coder, the 20 milliseconds of speech uses 20 bytes of the audio packet. There are an additional 40 bytes of the audio packet used for overhead, 20 bytes for an IP header, 8 bytes for an UDP header, and 12 bytes for a RTP header. The overhead to payload ratio is then 2 to 1, with two bytes of packet header for every one byte of audio packet payload.
- When the packet network is congested, it is important to use network bandwidth efficiently. When there is too much congestion, a network processing node may drop some of the transmitted packets. Depending upon the speech encoding algorithm used in the audio encoder, the sound quality of the audio signal degenerates rapidly as more packets are discarded. The large overhead required for transmitting a data stream over the packet network substantially increases this network congestion causing more packets to be delayed or even dropped, in turn, reducing the quality of data transmitted over the packet network.
- Accordingly, a need remains for a system that uses network bandwidth more effectively to improve transmission quality of data streams in a packet network.
- The size of packet payloads are dynamically adapted to the amount of congestion in a packet network. More data is put in packet payloads when more congestion exists in the packet network. When network congestion is high, less network bandwidth is available for transmitting packets. Accordingly, the packets are transmitted with larger payloads. When there is little or no network congestion smaller packet payloads are transmitted. The additional overhead created in transmitting smaller packets is acceptable when there is little or no network congestion because the network has excess bandwidth. When the network is congested, this excess bandwidth no longer exists. Thus, more payload is loaded into each packet to reduce the overhead to payload ratio and, in turn, reduce bandwidth consumption. Thus, the packet payloads are dynamically adjusted to use network resources more effectively. Some users may be willing to trade off the delay inherent in packing more frames into a packet for increased efficiency.
- Data is transmitted over the packet network by first encoding a data stream into encoded data. The encoded data is converted by a packetizer into packets having a packet header and a packet payload. The packetizer transmits the packets over the packet network to a receiving endpoint while monitoring congestion in the packet network.
- In one embodiment of the invention, the data stream is an audio or video data stream generated by a telephone. The packetizer packetizes the encoded audio data into audio packets having a header and an audio payload. The size of the audio payload is increased by packing more audio frames into each audio packet. The size of audio payloads is then decreased when the packet network is no longer congested. Congestion is detected by measuring end-to-end delay between a transmitting gateway and a receiving gateway using an existing protocol such as RTCP.
- The foregoing and other objects, features and advantages of the invention will become more readily apparent from the following detailed description of a preferred embodiment of the invention which proceeds with reference to the accompanying drawings.
- FIG. 1 is a schematic diagram of a packet telephony system that dynamically varies the size of audio packets according to network congestion.
- FIG. 2 is a schematic diagram of a transmitting gateway used in the packet telephony system shown in FIG. 1.
- FIG. 3 is a schematic diagram of a receiving gateway used in the packet telephony system shown in FIG. 1.
- FIG. 4 is a schematic diagram of variable sized packet payloads transmitted by the transmitting gateway shown in FIG. 2.
- FIG. 5 is a flow diagram describing how a packetizer in the transmitting gateway shown in FIG. 2 operates.
- FIG. 6 is a graph showing network bandwidth consumption for different header to payload ratios.
- FIG. 1 shows the general topology of a
packet telephony system 12 that varies the size of packet payloads according to measured network congestion. It should be understood that the invention is applicable to any application where streaming or real-time data is packetized for transmission over a packet network. For example, the invention is equally applicable to video streams or multimedia data streams. - The
packet telephony system 12 includesmultiple telephone handsets 14 connected to apacket network 16 throughgateways 18. Thepacket gateways 18 each include a codec for converting audio signals into audio packets and converting the audio packets back into audio signals. Thehandsets 14 are traditional telephones.Gateways 18 and the codecs used by thegateways 18 are any one of a wide variety of commercially available devices used for connecting thehandsets 14 to thepacket network 16. For example, thegateways 18 can be Voice Over Internet Protocol (VoIP) telephones or personal computers that include a digital signal processor (DSP) and software for encoding audio signals into audio packets. - The
gateways 18 operate as a transmitting gateway when encoding audio signals into audio packets and transmitting the audio packets over thepacket network 16 to a receiving gateway. Thegateways 18 operate as the receiving gateway when receiving audio packets over thepacket network 16 and decoding the audio packets back into audio signals. - A gateway transmit path is shown in the transmitting
packet gateway 20 in FIG. 2. The transmittingpacket gateway 20 includes a voice encoder 22, apacketizer 24, and atransmitter 26. Voice encoder 22 implements the compression half of a codec.Packetizer 24 accepts compressed audio data from encoder 22 and formats the data into packets for transmission. Thepacketizer 24 receives an end-to-end delay signal 25 back frompacket network 16. The end-to-end delay signal 25 is generated in various ways such as from a Real Time Protocol (RTP) report sent back from a receivingpacket gateway 28 shown in FIG. 3. Atransmitter 26 places the audio packets frompacketizer 24 ontopacket network 16. - The
receiving packet gateway 28 is shown in FIG. 3. Thereceiving gateway 28 reverses the process in transmittinggateway 20. Adepacketizer 30 accepts packets frompacket network 18 and separates out the audio frames. Ajitter buffer 32 buffers the audio frames and outputs them to avoice decoder 34 in an orderly manner. Thevoice decoder 34 implements the decompression half of the codec employed by voice encoder 22 (FIG. 2). The decoded audio frames are then output to telephone 14. The operations necessary to transmit and receive audio packets performed by the voice encoder 22,decoder 34,transmitter 26,packetizer 24 anddepacketizer 30 are well known and, therefore, not described in further detail. - Referring back to FIG. 1, an end-to-end packet delay11 is used to identify congestion occurring at any point in the
packet network 16. Congestion is defined as heavy network utilization experienced by one or more network processing elements such asrouters 19 and/orpacket gateways 18. Congested network processing element(s) can “back-up”, delaying processing and routing ofpackets 13 through thepacket network 16. If the congestion is severe, packets may be discarded by one or more of the network processing elements. - To reduce congestion, the overhead to payload ratio between a
packet header 15 and apacket payload 17 in thepacket 13 is adapted to the current congestion conditions inpacket network 16. When there is little or no congestion on thepacket network 16, asmaller packet payload 17 is packed into eachvoice packet 13. The delay in transmitting theaudio packet 13 is, in turn, shorter because the transmittinggateway 20 encodes and transmits a shorter portion of anaudio stream 10 output from one oftelephones 14. - When the
packet network 16 is congested, the transmittinggateway 20 increases the amount of audio data (payload) 17 as shown inaudio packet 21. The audio payload is dynamically increased while keepingheader 15 the same size. Less network bandwidth is used to transmit theaudio stream 10 because more audio data is transmitted using the same amount ofpacket overhead 15. This reduces congestion on thepacket network 16 and reduces the likelihood of packets being dropped or further delayed. - Network congestion is inferred by the amount of time it takes the audio packets to travel between the transmitting
gateway 20 and the receivinggateway 28. This end-to-end delay 11 is calculated using existing packet based voice protocols, such as Real Time Protocol (RTP RFC 1889) and Real Time Control Protocol (RTCP). RTP provides end-to-end transport for applications of streaming or real-time data, such as audio or video. RTCP provides estimates of network performance. - RTP and RTCP enable the receiving gateway to synchronize the received packets in the proper order so the user hears or sees the information correctly. Logical framing defines how the protocol “frames” or packages the audio or video data into bits (packets) for transport over a selected communications channel. Sequence numbering determines the order of data packets transported over a communications channel. RTCP also contains a system for determining end-to-end delay and periodically reporting that end-to-end delay back to the transmitting
gateway 20. Any other dynamic measure of end-to-end delay or network congestion can similarly be used as an congestion identifier topacketizer 24. - Referring to FIG. 4, the network end-to-end11 delay provided with the RTCP report is used by the
packetizer 24 to automatically vary the number of audio frames placed in each packet payload. This amount of audio data typically varies from 10-20 ms up to some maximum such as 100 ms. However, smaller or larger audio payloads may be used depending on specific network conditions. - The
audio packets packet network 16 using an Internet Protocol (IP). The audio packets include an IP header that is 20 bytes long, a User Datagram Protocol (UDP) header that is 8 bytes long, an RTP header that is 12 bytes long, and a variable sized audio payload. With little or no network congestion, usually 20 ms of speech are packed intoaudio packet 40. The 20 ms of speech is encoded into approximately 20 bytes of packet payload. The 40 bytes of overhead including the IP header, UDP header, and RTP header inpacket 40 takes up two thirds ofaudio packet 40. Every 20 ms. (50 times per second) a 60byte packet 40 is then generated and transmitted by transmitting gateway 20 (FIG. 2). - When there is medium congestion in the
packet network 16, audio packets similar topacket 42 are generated by the packetizer 24 (FIG. 2). Thepacket 42 carries 40 ms of audio data in a 40 byte packet payload but still uses only 40 bytes of overhead. The overhead ratio for transmitting 40 ms of speech is thereby reduced to one half of the total size ofpacket 42 at the cost of a 40 ms delay. - If heavy congestion is detected on the
packet network 16, thepacketizer 24 generates audio packets similar topacket 44.Packet 44 has a still larger audio payload of 100 ms. or more. The overhead ratio for transmitting 100 ms of speech is reduced further to one fifth of the total size ofpacket 44. - It should be noted that the amount of audio data in each packet is varied independently of the audio encoder22 (FIG. 22). Thus, the encoding scheme used to encode and decode the audio data does not have to be changed for different packet network conditions. This reduces encoder complexity. Because the size of audio packets and audio packet payloads is relayed in the packet header information, no modifications have to be made to existing network transport protocols. There are several well known algorithms for performing real-time adaptation that can be applied here. FIG. 5 demonstrates one, but the central idea of this invention does not rely on any specific adaptation algorithm.
- FIG. 5 is a flow diagram showing in more detail how the
packetizer 24 in FIG. 2 operates. The packetizer 22 is initialized for a given packet payload size instep 46. Thepacketizer 24 instep 48 packetizes encoded data from voice encoder 22 at the selected packet payload size. While packets are output bytransmitter 26, thepacketizer 24 instep 50 monitors thepacket network 16 for congestion. Decision step 52 determines whether the current packet payload size is within a range compatible with the current network congestion condition. This is can be done using a table previously loaded into thepacketizer 24. The table contains acceptable packet payload sizes for different end-to-end network delays. - If the payload size is within range, the
packetizer 24 jumps back to step 48 and continues to packetize audio data at the current payload size. If the current payload size is not within an acceptable range for the current network congestion,decision step 54 determines whether the current packet payload is either too small or too large. -
Decision step 54 decides whether the packet payload size is too small for the current end-to-end delay. If so, thepacketizer 24 automatically increases the audio packet payload size instep 56. If the packet payload is too large, the audio packet payload size is automatically decreased by thepacketizer 24 instep 58. The packetizer then jumps back to step 48 and packetizes audio data at the new packet payload size. - FIG. 6 is a graph showing bandwidth consumption in a packet network for different header to payload ratios. Each line represents a different codec bit rates. This graph can be used as a reference in
packetizer 24 for changing the packet payload size. - The invention dynamically changes the overhead to packet payload ratio to more effectively adapt to current network congestion conditions. By improving network bandwidth efficiency, the quality of streaming and real-time data transmitted over the packet network is improved.
- Having described and illustrated the principles of the invention in a preferred embodiment thereof, it should be apparent that the invention can be modified in arrangement and detail without departing from such principles. I claim all modifications and variation coming within the spirit and scope of the following claims.
Claims (14)
1. A system for transmitting packets over a packet network, comprising:
an encoder for encoding a data stream;
a packetizer coupled to the encoder converting the encoded data stream into packets each having a packet header and a packet payload, the packetizer monitoring congestion in the packet network and dynamically varying the size of the packet payload in the packets according to an amount of monitored congestion, thus trading off packetization delay for network efficiency.
2. A system according to wherein congestion in the packet network is monitored by measuring end-to-end delay of the packets between an originating endpoint containing the packetizer and a destination endpoint for the packets.
claim 1
3. A system according to wherein the end-to-end delay is provided to the packetizer using a RTCP report.
claim 1
4. A system according to wherein the packet header remains at substantially the same size regardless of the amount of congestion in the packet network.
claim 1
5. A system according to wherein the data stream comprises an audio stream encoded by the encoder and more encoded audio data is packed into each packet payload when there is more congestion in the packet network and less encoded audio data is packed into each packet payload when there is less congestion in the packet network.
claim 1
6. A system according to wherein the packet header includes an IP header, an UDP header and a RTP header.
claim 5
7. A gateway according to wherein the encoder is coupled to a telephone that generates the audio stream.
claim 5
8. A method for transmitting data over a packet network, comprising:
encoding a data stream into encoded data;
converting the encoded data into packets having packet headers and packet payloads;
transmitting the packets over the packet network to a receiving endpoint while detecting congestion in the packet network; and
automatically increasing a size of the packet payloads in the transmitted packets when congestion is detected in the packet network to reduce a percentage of the transmitted packets used as packet overhead.
9. A method according to wherein detecting congestion comprises measuring end-to-end delay between a packet transmitting endpoint and a packet receiving endpoint.
claim 8
10. A method according to wherein the end-to-end delay is measured using a RTP.
claim 9
11. A method according to wherein the data stream is an audio data stream generated from a telephone and the encoded audio data is packetized into audio packets having an audio header and an audio payload.
claim 8
12. A method according to including increasing the audio payload by delaying the transmission of each packet to encode and pack a larger amount of audio data into the packets and decreasing the amount of audio payload in the packets when the packet network is no longer congested.
claim 11
13. A method according to including using about 40 bytes for the audio header and about 20 bytes for the audio payload in each packet when there is little or no network congestion and using about 40 bytes for the packet header and about 40 or more bytes for the audio payload when there is greater congestion in the packet network.
claim 12
14. A system for transmitting audio packets over a packet network, comprising:
a transmitting gateway having an encoder coupled to a telephone for encoding an audio stream into speech frames, a packetizer coupled to the encoder formatting the speech frames into audio packets each having packet headers and packet payloads including one or more of the speech frames, and a transmitter coupled between the packetizer and the packet network for transmitting the audio packets over the packet network; and
a receiving gateway having a depacketizer for depacketizing the audio packet payloads into speech frames, a jitter buffer delaying decoding of the speech frames to account for variances in audio packet delays and a voice decoder for decoding the speech frames back into a decoded audio stream,
the packetizer in the transmitting gateway dynamically varying the number of speech frames in the audio packet payloads according to audio packet transmission delays in the packet network between the transmitting gateway and the receiving gateway.
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US10/142,232 US6886040B1 (en) | 1998-10-28 | 2002-05-08 | Codec-independent technique for modulating bandwidth in packet network |
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US09/181,947 US6421720B2 (en) | 1998-10-28 | 1998-10-28 | Codec-independent technique for modulating bandwidth in packet network |
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Cited By (34)
Publication number | Priority date | Publication date | Assignee | Title |
---|---|---|---|---|
EP1372300A1 (en) * | 2002-06-10 | 2003-12-17 | Alcatel | Adapting packet length to network load for VoIP communications |
EP1376947A1 (en) * | 2002-06-27 | 2004-01-02 | Siemens Aktiengesellschaft | Anpassung der Zahl der pro Paket übertragenen Rahmen an die Netzwerklast |
US20040022203A1 (en) * | 2002-07-30 | 2004-02-05 | Michelson Steven M. | Method of sizing packets for routing over a communication network for VoIP calls on a per call basis |
US6856613B1 (en) * | 1999-12-30 | 2005-02-15 | Cisco Technology, Inc. | Method and apparatus for throttling audio packets according to gateway processing capacity |
US20050068893A1 (en) * | 2001-02-14 | 2005-03-31 | Microsoft Corporation | System for refining network utilization and data block sizes in the transfer of data over a network |
US6888793B1 (en) * | 1999-06-01 | 2005-05-03 | Nec Corporation | Internet protocol network alternate routing system |
US20060045073A1 (en) * | 2004-08-31 | 2006-03-02 | Telefonaktiebolaget Lm Ericsson (Publ) | Frame size adaptation in real-time transport protocol |
WO2006025789A1 (en) * | 2004-08-31 | 2006-03-09 | Telefonaktiebolaget Lm Ericsson (Publ) | Method and system for frame size adaptation in real-time transport protocol |
US20060088032A1 (en) * | 2004-10-26 | 2006-04-27 | Bradley Venables | Method and system for flow management with scheduling |
US7069342B1 (en) * | 2001-03-01 | 2006-06-27 | Cisco Technology, Inc. | Communication system with content-based data compression |
US20060171446A1 (en) * | 2005-01-31 | 2006-08-03 | Nec Electronics Corporation | Data multiplexing apparatus |
EP1727330A1 (en) * | 2005-05-25 | 2006-11-29 | Kyocera Corporation | Wireless Communication Method and Apparatus |
US20070094409A1 (en) * | 2005-10-20 | 2007-04-26 | Crockett Douglas M | System and method for adaptive media bundling for voice over internet protocol applications |
US20070147314A1 (en) * | 2005-12-22 | 2007-06-28 | Telefonaktiebolaget Lm Ericsson (Publ) | Network processing node and method for manipulating packets |
EP1901495A1 (en) | 2006-09-13 | 2008-03-19 | Broadcom Corporation | Adaptive packet size modification for voice over packet networks |
US20080181259A1 (en) * | 2007-01-31 | 2008-07-31 | Dmitry Andreev | Method and system for dynamically adjusting packet size to decrease delays of streaming data transmissions on noisy transmission lines |
US7437428B1 (en) | 2000-02-16 | 2008-10-14 | Microsoft Corporation | System and method for transferring data over a network |
US20100008430A1 (en) * | 2008-07-11 | 2010-01-14 | Qualcomm Incorporated | Filtering video data using a plurality of filters |
US20100177822A1 (en) * | 2009-01-15 | 2010-07-15 | Marta Karczewicz | Filter prediction based on activity metrics in video coding |
US20100198969A1 (en) * | 2000-08-24 | 2010-08-05 | Aol Llc | Deep Packet Scan Hacker Identification |
US20100232349A1 (en) * | 2009-03-11 | 2010-09-16 | Xcast Labs, Inc. | Optimizing voip for satellite connection |
US20110122875A1 (en) * | 2000-05-30 | 2011-05-26 | Juniper Networks, Inc. | Sts frame-atm cell circuit emulation apparatus and frame length compensation method for the same |
US8098686B1 (en) * | 2005-12-02 | 2012-01-17 | At&T Intellectual Property Ii, L.P. | Method and apparatus for providing an application-level utility metric |
WO2012047304A1 (en) * | 2010-10-04 | 2012-04-12 | Vidyo, Inc. | Delay aware rate control in the context of hierarchical p picture coding |
CN102547243A (en) * | 2012-01-17 | 2012-07-04 | 西安电子科技大学 | Audio and video remote monitoring method and system based on 3G (the 3rd generation telecommunication) network |
WO2012123762A1 (en) * | 2011-03-17 | 2012-09-20 | Bae Systems Plc | Improvements in call delay control |
US8276035B1 (en) * | 2008-07-28 | 2012-09-25 | Netmotion Wireless, Inc. | High performance digital communications resiliency in a roamable virtual private network |
EP2503746A1 (en) * | 2011-03-25 | 2012-09-26 | Fujitsu Limited | Data transfer apparatus, data transfer method, and information processing apparatus |
EP2755343A1 (en) * | 2013-01-15 | 2014-07-16 | S.A.V.E.T. Srl | System and mobile terminal using data fragmentation |
US8964852B2 (en) | 2011-02-23 | 2015-02-24 | Qualcomm Incorporated | Multi-metric filtering |
US20150180932A1 (en) * | 2013-12-23 | 2015-06-25 | David Arthur Yost | System for intelligible audio conversation over unreliable digital transmission media |
WO2018039015A1 (en) * | 2016-08-24 | 2018-03-01 | Microsoft Technology Licensing, Llc | Media buffering |
US10270703B2 (en) | 2016-08-23 | 2019-04-23 | Microsoft Technology Licensing, Llc | Media buffering |
US11336584B2 (en) * | 2016-12-07 | 2022-05-17 | Fuji Corporation | Communication control device that varies data partitions based on a status of connected nodes |
Families Citing this family (50)
Publication number | Priority date | Publication date | Assignee | Title |
---|---|---|---|---|
WO1999044363A1 (en) * | 1998-02-27 | 1999-09-02 | Ridgeway Systems And Software Ltd. | Audio-video packet synchronisation at network gateway |
US6421720B2 (en) * | 1998-10-28 | 2002-07-16 | Cisco Technology, Inc. | Codec-independent technique for modulating bandwidth in packet network |
US6529475B1 (en) * | 1998-12-16 | 2003-03-04 | Nortel Networks Limited | Monitor for the control of multimedia services in networks |
US6678250B1 (en) * | 1999-02-19 | 2004-01-13 | 3Com Corporation | Method and system for monitoring and management of the performance of real-time networks |
US6996059B1 (en) * | 1999-05-19 | 2006-02-07 | Shoretel, Inc | Increasing duration of information in a packet to reduce processing requirements |
US6697871B1 (en) * | 1999-09-21 | 2004-02-24 | Network Associates Technology, Inc. | System and method for efficient encoding and decoding of protocol messages |
EP1087579B1 (en) * | 1999-09-22 | 2004-12-01 | NTT DoCoMo, Inc. | Gateway for reducing delay jitter and method for data transfer therein |
US6650652B1 (en) * | 1999-10-12 | 2003-11-18 | Cisco Technology, Inc. | Optimizing queuing of voice packet flows in a network |
US6889257B1 (en) * | 1999-12-03 | 2005-05-03 | Realnetworks, Inc. | System and method of transmitting data packets |
US6785234B1 (en) * | 1999-12-22 | 2004-08-31 | Cisco Technology, Inc. | Method and apparatus for providing user control of audio quality |
US6625124B1 (en) * | 2000-03-03 | 2003-09-23 | Luminous Networks, Inc. | Automatic reconfiguration of short addresses for devices in a network due to change in network topology |
US7111058B1 (en) * | 2000-06-28 | 2006-09-19 | Cisco Technology, Inc. | Server and method for transmitting streaming media to client through a congested network |
JP3584859B2 (en) * | 2000-06-29 | 2004-11-04 | 日本電気株式会社 | Packet scheduling device |
US20020075869A1 (en) * | 2000-12-18 | 2002-06-20 | Shah Tushar Ramanlal | Integration of network, data link, and physical layer to adapt network traffic |
US7333512B2 (en) * | 2000-12-18 | 2008-02-19 | Rmi Corporation | Dynamic mixing TDM data with data packets |
US7787447B1 (en) * | 2000-12-28 | 2010-08-31 | Nortel Networks Limited | Voice optimization in a network having voice over the internet protocol communication devices |
AU2002306749A1 (en) * | 2001-03-13 | 2002-09-24 | Shiv Balakrishnan | An architecture and protocol for a wireless communication network to provide scalable web services to mobile access devices |
JP2002300274A (en) * | 2001-03-30 | 2002-10-11 | Fujitsu Ltd | Gateway device and voice data transfer method |
US7089320B1 (en) * | 2001-06-01 | 2006-08-08 | Cisco Technology, Inc. | Apparatus and methods for combining data |
US9836424B2 (en) * | 2001-08-24 | 2017-12-05 | Intel Corporation | General input/output architecture, protocol and related methods to implement flow control |
KR100750036B1 (en) * | 2001-08-24 | 2007-08-16 | 인텔 코오퍼레이션 | A general input/output architecture, protocol and related methods to implement flow control |
AU2003303131A1 (en) * | 2002-07-26 | 2004-07-29 | Network General Technology | Network analyzer co-processor system and method |
KR100663586B1 (en) * | 2002-08-28 | 2007-01-02 | 삼성전자주식회사 | Method and apparatus transmitting a header compressed packet data |
FR2849980B1 (en) * | 2003-01-15 | 2005-04-08 | Medialive | METHOD FOR THE DISTRIBUTION OF VIDEO SEQUENCES, DECODER AND SYSTEM FOR THE IMPLEMENTATION OF THIS PRODUCT |
EP1447953B1 (en) * | 2003-02-11 | 2005-01-19 | Alcatel | Gateway |
JP2004260658A (en) * | 2003-02-27 | 2004-09-16 | Matsushita Electric Ind Co Ltd | Wireless lan device |
FR2853786B1 (en) * | 2003-04-11 | 2005-08-05 | Medialive | METHOD AND EQUIPMENT FOR DISTRIBUTING DIGITAL VIDEO PRODUCTS WITH A RESTRICTION OF CERTAIN AT LEAST REPRESENTATION AND REPRODUCTION RIGHTS |
FR2854530B1 (en) * | 2003-05-02 | 2005-07-22 | Medialive | METHOD AND DEVICE FOR SECURING THE TRANSMISSION, RECORDING AND VISUALIZATION OF DIGITAL AUDIOVISUAL EMPTY STREAMS |
US7310480B2 (en) * | 2003-06-18 | 2007-12-18 | Intel Corporation | Adaptive framework for closed-loop protocols over photonic burst switched networks |
FR2858899B1 (en) * | 2003-08-11 | 2005-12-02 | Medialive | SECURE DISTRIBUTED METHOD AND SYSTEM FOR AUDIOVISUAL FLOW PROTECTION AND DISTRIBUTION |
GB2405773B (en) * | 2003-09-02 | 2006-11-08 | Siemens Ag | A method of controlling provision of audio communication on a network |
US20050068968A1 (en) * | 2003-09-30 | 2005-03-31 | Shlomo Ovadia | Optical-switched (OS) network to OS network routing using extended border gateway protocol |
FR2861240B1 (en) * | 2003-10-15 | 2006-03-03 | Medialive | SECURE DISTRIBUTED METHOD AND SYSTEM FOR AUDIOVISUAL FLOW DISTRIBUTION |
WO2005043178A2 (en) * | 2003-10-29 | 2005-05-12 | University Of Pittsburgh Of The Commonwealth System Of Higher Education | Optimizing packetization for minimal end-to-end delay in voip networks |
FR2862835B1 (en) * | 2003-11-24 | 2006-04-14 | Medialive | SECURED AND CUSTOMIZED DIFFUSION OF AUDIOVISUAL FLOWS BY A UNICAST / MULTICAST HYBRID SYSTEM |
US7734176B2 (en) * | 2003-12-22 | 2010-06-08 | Intel Corporation | Hybrid optical burst switching with fixed time slot architecture |
US20060015579A1 (en) * | 2004-07-16 | 2006-01-19 | Bharat Sastri | Architecture and protocol for a wireless communication network to provide scalable web services to mobile access devices |
US7773517B2 (en) * | 2004-11-19 | 2010-08-10 | Research In Motion Limited | Method and system for identifying degradation of a media service |
US7480238B2 (en) * | 2005-04-14 | 2009-01-20 | International Business Machines Corporation | Dynamic packet training |
US7768933B2 (en) * | 2005-10-14 | 2010-08-03 | Chang Kirk K | Estimating available bandwidth and enhancing narrow link bandwidth estimations in telecommunications networks using existing user traffic |
WO2008000289A1 (en) * | 2006-06-29 | 2008-01-03 | Telecom Italia S.P.A. | Method and apparatus for improving bandwith exploitation in real-time audio/video communications |
FI120858B (en) * | 2007-02-09 | 2010-03-31 | Teliasonera Ab | Transmission of real-time user data frames in packages |
US7987285B2 (en) * | 2007-07-10 | 2011-07-26 | Bytemobile, Inc. | Adaptive bitrate management for streaming media over packet networks |
US9398490B2 (en) * | 2013-03-15 | 2016-07-19 | Trane International Inc. | Method of fragmenting a message in a network |
US9473363B2 (en) * | 2013-07-15 | 2016-10-18 | Globalfoundries Inc. | Managing quality of service for communication sessions |
US9491076B2 (en) | 2014-01-06 | 2016-11-08 | Cisco Technology, Inc. | Learning end-to-end delays in computer networks from sporadic round-trip delay probing |
US9774522B2 (en) | 2014-01-06 | 2017-09-26 | Cisco Technology, Inc. | Triggering reroutes using early learning machine-based prediction of failures |
US10277476B2 (en) | 2014-01-06 | 2019-04-30 | Cisco Technology, Inc. | Optimizing network parameters based on a learned network performance model |
US9338065B2 (en) | 2014-01-06 | 2016-05-10 | Cisco Technology, Inc. | Predictive learning machine-based approach to detect traffic outside of service level agreements |
US10269199B2 (en) | 2017-09-27 | 2019-04-23 | Honda Motor Co., Ltd. | System and method for providing energy efficient hands free vehicle door operation |
Family Cites Families (25)
Publication number | Priority date | Publication date | Assignee | Title |
---|---|---|---|---|
US4679227A (en) | 1985-05-20 | 1987-07-07 | Telebit Corporation | Ensemble modem structure for imperfect transmission media |
US4771391A (en) * | 1986-07-21 | 1988-09-13 | International Business Machines Corporation | Adaptive packet length traffic control in a local area network |
US4769815A (en) * | 1987-04-10 | 1988-09-06 | American Telephone And Telegraph Company, At&T Bell Laboratories | Packet flow control method |
US5231633A (en) * | 1990-07-11 | 1993-07-27 | Codex Corporation | Method for prioritizing, selectively discarding, and multiplexing differing traffic type fast packets |
US5115429A (en) * | 1990-08-02 | 1992-05-19 | Codex Corporation | Dynamic encoding rate control minimizes traffic congestion in a packet network |
US5384770A (en) * | 1992-05-08 | 1995-01-24 | Hayes Microcomputer Products, Inc. | Packet assembler |
US5506844A (en) * | 1994-05-20 | 1996-04-09 | Compression Labs, Inc. | Method for configuring a statistical multiplexer to dynamically allocate communication channel bandwidth |
US5541919A (en) * | 1994-12-19 | 1996-07-30 | Motorola, Inc. | Multimedia multiplexing device and method using dynamic packet segmentation |
US6411725B1 (en) * | 1995-07-27 | 2002-06-25 | Digimarc Corporation | Watermark enabled video objects |
US5826032A (en) * | 1996-02-12 | 1998-10-20 | University Of Southern California | Method and network interface logic for providing embedded checksums |
US6002669A (en) * | 1996-03-26 | 1999-12-14 | White; Darryl C. | Efficient, multi-purpose network data communications protocol |
US6298057B1 (en) * | 1996-04-19 | 2001-10-02 | Nortel Networks Limited | System and method for reliability transporting aural information across a network |
US5764645A (en) * | 1996-06-12 | 1998-06-09 | Microsoft Corporation | IP/ATM network adaptation |
US6005871A (en) * | 1996-08-22 | 1999-12-21 | Telefonaktiebolaget Lm Ericsson | Minicell alignment |
US5940479A (en) * | 1996-10-01 | 1999-08-17 | Northern Telecom Limited | System and method for transmitting aural information between a computer and telephone equipment |
US6075770A (en) * | 1996-11-20 | 2000-06-13 | Industrial Technology Research Institute | Power spectrum-based connection admission control for ATM networks |
US6304567B1 (en) * | 1996-11-26 | 2001-10-16 | Lucent Technologies Inc. | Methods and apparatus for providing voice communications through a packet network |
US6003089A (en) * | 1997-03-31 | 1999-12-14 | Siemens Information And Communication Networks, Inc. | Method for constructing adaptive packet lengths in a congested network |
US6006253A (en) * | 1997-10-31 | 1999-12-21 | Intel Corporation | Method and apparatus to provide a backchannel for receiver terminals in a loosely-coupled conference |
US6493343B1 (en) * | 1998-01-07 | 2002-12-10 | Compaq Information Technologies Group | System and method for implementing multi-pathing data transfers in a system area network |
US6370163B1 (en) * | 1998-03-11 | 2002-04-09 | Siemens Information And Communications Network, Inc. | Apparatus and method for speech transport with adaptive packet size |
US6466586B1 (en) * | 1998-03-31 | 2002-10-15 | Nortel Networks Limited | Digital subscriber line framing structure supporting imbedded rate adaptive synchronous and asynchronous traffic |
US6052368A (en) * | 1998-05-22 | 2000-04-18 | Cabletron Systems, Inc. | Method and apparatus for forwarding variable-length packets between channel-specific packet processors and a crossbar of a multiport switch |
US6728263B2 (en) * | 1998-08-18 | 2004-04-27 | Microsoft Corporation | Dynamic sizing of data packets |
US6421720B2 (en) * | 1998-10-28 | 2002-07-16 | Cisco Technology, Inc. | Codec-independent technique for modulating bandwidth in packet network |
-
1998
- 1998-10-28 US US09/181,947 patent/US6421720B2/en not_active Expired - Lifetime
-
2002
- 2002-05-08 US US10/142,232 patent/US6886040B1/en not_active Expired - Fee Related
Cited By (72)
Publication number | Priority date | Publication date | Assignee | Title |
---|---|---|---|---|
US6888793B1 (en) * | 1999-06-01 | 2005-05-03 | Nec Corporation | Internet protocol network alternate routing system |
US20050122985A1 (en) * | 1999-12-30 | 2005-06-09 | Cisco Technology, Inc. | Method and apparatus for throttling audio packets according to gateway processing capacity |
US7567576B2 (en) | 1999-12-30 | 2009-07-28 | Cisco Technology, Inc. | Method and apparatus for throttling audio packets according to gateway processing capacity |
US6856613B1 (en) * | 1999-12-30 | 2005-02-15 | Cisco Technology, Inc. | Method and apparatus for throttling audio packets according to gateway processing capacity |
US7437428B1 (en) | 2000-02-16 | 2008-10-14 | Microsoft Corporation | System and method for transferring data over a network |
US20110122875A1 (en) * | 2000-05-30 | 2011-05-26 | Juniper Networks, Inc. | Sts frame-atm cell circuit emulation apparatus and frame length compensation method for the same |
US8619815B2 (en) * | 2000-05-30 | 2013-12-31 | Juniper Networks, Inc. | Frame length compensation |
US20100198969A1 (en) * | 2000-08-24 | 2010-08-05 | Aol Llc | Deep Packet Scan Hacker Identification |
US8001244B2 (en) * | 2000-08-24 | 2011-08-16 | Aol Inc. | Deep packet scan hacker identification |
US8645537B2 (en) | 2000-08-24 | 2014-02-04 | Citrix Systems, Inc. | Deep packet scan hacker identification |
US20050068892A1 (en) * | 2001-02-14 | 2005-03-31 | Microsoft Corporation | Method for transferring data over a network |
US7522536B2 (en) | 2001-02-14 | 2009-04-21 | Microsoft Corporation | Method for transferring data over a network |
US20050091398A1 (en) * | 2001-02-14 | 2005-04-28 | Microsoft Corporation | System for transferring data over a network |
US7436771B2 (en) | 2001-02-14 | 2008-10-14 | Microsoft Corporation | System for refining network utilization and data block sizes in the transfer of data over a network |
US20050068893A1 (en) * | 2001-02-14 | 2005-03-31 | Microsoft Corporation | System for refining network utilization and data block sizes in the transfer of data over a network |
US20050091397A1 (en) * | 2001-02-14 | 2005-04-28 | Microsoft Corporation | Method and system for managing data transfer over a network |
US7502849B2 (en) | 2001-02-14 | 2009-03-10 | Microsoft Corporation | System for transferring data over a network |
US7325068B2 (en) * | 2001-02-14 | 2008-01-29 | Microsoft Corporation | Method and system for managing data transfer over a network |
US7069342B1 (en) * | 2001-03-01 | 2006-06-27 | Cisco Technology, Inc. | Communication system with content-based data compression |
EP1372300A1 (en) * | 2002-06-10 | 2003-12-17 | Alcatel | Adapting packet length to network load for VoIP communications |
EP1376947A1 (en) * | 2002-06-27 | 2004-01-02 | Siemens Aktiengesellschaft | Anpassung der Zahl der pro Paket übertragenen Rahmen an die Netzwerklast |
WO2004004243A1 (en) * | 2002-06-27 | 2004-01-08 | Siemens Aktiengesellschaft | Matching of the number of frames transmitted per packet to the network load |
US20040022203A1 (en) * | 2002-07-30 | 2004-02-05 | Michelson Steven M. | Method of sizing packets for routing over a communication network for VoIP calls on a per call basis |
US20080031229A1 (en) * | 2002-07-30 | 2008-02-07 | Michelson Steven M | Method of sizing packets for routing over a communication network for voip calls on a per call basis |
US7283541B2 (en) * | 2002-07-30 | 2007-10-16 | At&T Corp. | Method of sizing packets for routing over a communication network for VoIP calls on a per call basis |
US8199762B2 (en) | 2002-07-30 | 2012-06-12 | At&T Intellectual Property Ii, L.P. | Method of sizing packets for routing over a communication network for VoIP calls on a per call basis |
WO2006025789A1 (en) * | 2004-08-31 | 2006-03-09 | Telefonaktiebolaget Lm Ericsson (Publ) | Method and system for frame size adaptation in real-time transport protocol |
US20060045073A1 (en) * | 2004-08-31 | 2006-03-02 | Telefonaktiebolaget Lm Ericsson (Publ) | Frame size adaptation in real-time transport protocol |
US7672272B2 (en) * | 2004-08-31 | 2010-03-02 | Telefonaktiebolaget L M Ericsson (Publ) | Frame size adaptation in real-time transport protocol |
US20060088032A1 (en) * | 2004-10-26 | 2006-04-27 | Bradley Venables | Method and system for flow management with scheduling |
US8249433B2 (en) * | 2005-01-31 | 2012-08-21 | Renesas Electronics Corporation | Data multiplexing apparatus |
US20060171446A1 (en) * | 2005-01-31 | 2006-08-03 | Nec Electronics Corporation | Data multiplexing apparatus |
KR100849290B1 (en) * | 2005-05-25 | 2008-07-29 | 교세라 가부시키가이샤 | Wireless communication method and apparatus |
EP1727330A1 (en) * | 2005-05-25 | 2006-11-29 | Kyocera Corporation | Wireless Communication Method and Apparatus |
US20060268865A1 (en) * | 2005-05-25 | 2006-11-30 | Kyocera Corporation | Wireless communication method and apparatus |
US20070094409A1 (en) * | 2005-10-20 | 2007-04-26 | Crockett Douglas M | System and method for adaptive media bundling for voice over internet protocol applications |
US8098686B1 (en) * | 2005-12-02 | 2012-01-17 | At&T Intellectual Property Ii, L.P. | Method and apparatus for providing an application-level utility metric |
US20070147314A1 (en) * | 2005-12-22 | 2007-06-28 | Telefonaktiebolaget Lm Ericsson (Publ) | Network processing node and method for manipulating packets |
EP1901495A1 (en) | 2006-09-13 | 2008-03-19 | Broadcom Corporation | Adaptive packet size modification for voice over packet networks |
US8391166B2 (en) | 2006-09-13 | 2013-03-05 | Broadcom Corporation | Adaptive packet size modification for voice over packet networks |
US8830865B2 (en) | 2006-09-13 | 2014-09-09 | Broadcom Corporation | Adaptive packet size modification for packet networks |
US7733785B2 (en) | 2007-01-31 | 2010-06-08 | International Business Machines Corporation | Method and system for dynamically adjusting packet size to decrease delays of streaming data transmissions on noisy transmission lines |
WO2008092784A1 (en) * | 2007-01-31 | 2008-08-07 | International Business Machines Corporation | Dynamically adjusting packet size to decrease delays of streaming data transmissions on noisy transmission lines |
US20080181259A1 (en) * | 2007-01-31 | 2008-07-31 | Dmitry Andreev | Method and system for dynamically adjusting packet size to decrease delays of streaming data transmissions on noisy transmission lines |
US20100008430A1 (en) * | 2008-07-11 | 2010-01-14 | Qualcomm Incorporated | Filtering video data using a plurality of filters |
US10123050B2 (en) | 2008-07-11 | 2018-11-06 | Qualcomm Incorporated | Filtering video data using a plurality of filters |
US11711548B2 (en) | 2008-07-11 | 2023-07-25 | Qualcomm Incorporated | Filtering video data using a plurality of filters |
US8276035B1 (en) * | 2008-07-28 | 2012-09-25 | Netmotion Wireless, Inc. | High performance digital communications resiliency in a roamable virtual private network |
US20100177822A1 (en) * | 2009-01-15 | 2010-07-15 | Marta Karczewicz | Filter prediction based on activity metrics in video coding |
US9143803B2 (en) | 2009-01-15 | 2015-09-22 | Qualcomm Incorporated | Filter prediction based on activity metrics in video coding |
US8488632B2 (en) * | 2009-03-11 | 2013-07-16 | Xcast Labs, Inc. | Optimizing VoIP for satellite connection |
US20100232349A1 (en) * | 2009-03-11 | 2010-09-16 | Xcast Labs, Inc. | Optimizing voip for satellite connection |
WO2012047304A1 (en) * | 2010-10-04 | 2012-04-12 | Vidyo, Inc. | Delay aware rate control in the context of hierarchical p picture coding |
US8750373B2 (en) | 2010-10-04 | 2014-06-10 | Vidyo, Inc. | Delay aware rate control in the context of hierarchical P picture coding |
US8964853B2 (en) | 2011-02-23 | 2015-02-24 | Qualcomm Incorporated | Multi-metric filtering |
US8964852B2 (en) | 2011-02-23 | 2015-02-24 | Qualcomm Incorporated | Multi-metric filtering |
US9877023B2 (en) | 2011-02-23 | 2018-01-23 | Qualcomm Incorporated | Multi-metric filtering |
US8982960B2 (en) | 2011-02-23 | 2015-03-17 | Qualcomm Incorporated | Multi-metric filtering |
US8989261B2 (en) | 2011-02-23 | 2015-03-24 | Qualcomm Incorporated | Multi-metric filtering |
US9258563B2 (en) | 2011-02-23 | 2016-02-09 | Qualcomm Incorporated | Multi-metric filtering |
US9819936B2 (en) | 2011-02-23 | 2017-11-14 | Qualcomm Incorporated | Multi-metric filtering |
WO2012123762A1 (en) * | 2011-03-17 | 2012-09-20 | Bae Systems Plc | Improvements in call delay control |
AU2012228036B2 (en) * | 2011-03-17 | 2015-09-17 | Bae Systems Plc | Improvements in call delay control |
US20140003231A1 (en) * | 2011-03-17 | 2014-01-02 | Bae Systems Plc | Call delay control |
EP2503746A1 (en) * | 2011-03-25 | 2012-09-26 | Fujitsu Limited | Data transfer apparatus, data transfer method, and information processing apparatus |
CN102547243A (en) * | 2012-01-17 | 2012-07-04 | 西安电子科技大学 | Audio and video remote monitoring method and system based on 3G (the 3rd generation telecommunication) network |
EP2755343A1 (en) * | 2013-01-15 | 2014-07-16 | S.A.V.E.T. Srl | System and mobile terminal using data fragmentation |
US9680905B2 (en) * | 2013-12-23 | 2017-06-13 | David Arthur Yost | System for intelligible audio conversation over unreliable digital transmission media |
US20150180932A1 (en) * | 2013-12-23 | 2015-06-25 | David Arthur Yost | System for intelligible audio conversation over unreliable digital transmission media |
US10270703B2 (en) | 2016-08-23 | 2019-04-23 | Microsoft Technology Licensing, Llc | Media buffering |
WO2018039015A1 (en) * | 2016-08-24 | 2018-03-01 | Microsoft Technology Licensing, Llc | Media buffering |
US11336584B2 (en) * | 2016-12-07 | 2022-05-17 | Fuji Corporation | Communication control device that varies data partitions based on a status of connected nodes |
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