Showing papers on "Packet loss published in 2001"

Delayed Internet routing convergence

Abstract: This paper examines the latency in Internet path failure, failover, and repair due to the convergence properties of interdomain routing. Unlike circuit-switched paths which exhibit failover on the order of milliseconds, our experimental measurements show that interdomain routers in the packet-switched Internet may take tens of minutes to reach a consistent view of the network topology after a fault. These delays stem from temporary routing table fluctuations formed during the operation of the border gateway protocol (BGP) path selection process on the Internet backbone routers. During these periods of delayed convergence, we show that end-to-end Internet paths will experience intermittent loss of connectivity, as well as increased packet loss and latency. We present a two-year study of Internet routing convergence through the experimental instrumentation of key portions of the Internet infrastructure, including both passive data collection and fault-injection machines at major Internet exchange points. Based on data from the injection and measurement of several hundred thousand interdomain routing faults, we describe several unexpected properties of convergence and show that the measured upper bound on Internet interdomain routing convergence delay is an order of magnitude slower than previously thought. Our analysis also shows that the upper theoretic computational bound on the number of router states and control messages exchanged during the process of BGP convergence is factorial with respect to the number of autonomous systems in the Internet. Finally, we demonstrate that much of the observed convergence delay stems from specific router vendor implementation decisions and ambiguity in the BGP specification.

Voice over IP performance monitoring

Abstract: We describe a method for monitoring Voice over IP (VoIP) applications based upon a reduction of the ITU-T's E-Model to transport level, measurable quantities. In the process, 1) we identify the relevant transport level quantities, 2) we discuss the tradeoffs between placing the monitors within the VoIP gateways versus placement of the monitors within the transport path, and 3) we identify several areas where further work and consensus within the industry are required. We discover that the relevant transport level quantities are the delay, network packet loss and the decoder's de-jitter buffer packet loss. We find that an in-path monitor requires the definition of a reference de-jitter buffer implementation to estimate voice quality based upon observed transport measurements. Finally, we suggest that more studies are required, which evaluate the quality of various VoIP codecs in the presence of representative packet loss patterns.

Supporting service differentiation in wireless packet networks using distributed control

Abstract: This paper investigates differentiated services in wireless packet networks using a fully distributed approach that supports service differentiation, radio monitoring, and admission control. While our proposal is generally applicable to distributed wireless access schemes, we design, implement, and evaluate our framework within the context of existing wireless technology. Service differentiation is based on the IEEE 802.11 distributed coordination function (DCF) originally designed to support best-effort data services. We analyze the delay experienced by a mobile host implementing the IEEE 802.11 DCF and derive a closed-form formula. We then extend the DCF to provide service differentiation for delay-sensitive and best-effort traffic based on the results from the analysis. Two distributed estimation algorithms are proposed. These algorithms are evaluated using simulation, analysis, and experimentation. A virtual MAC (VMAC) algorithm passively monitors the radio channel and estimates locally achievable service levels. The VMAC estimates key MAC level statistics related to service quality such as delay, delay variation, packet collision, and packet loss. We show the efficiency of the VMAC algorithm through simulation and consider significantly overlapping cells and highly bursty traffic mixes. In addition, we implement and evaluate the VMAC in an experimental differentiated services wireless testbed. A virtual source (VS) algorithm utilizes the VMAC to estimate application-level service quality. The VS allows application parameters to be tuned in response to dynamic channel conditions based on "virtual delay curves." We demonstrate through simulation that when these distributed victual algorithms are applied to the admission control of the radio channel then a globally stable state can be maintained without the need for complex centralized radio resource management.

Monitoring network activity

Abstract: A system for analysing network traffic, particularly to detect suspect packets and identify attacks or potential attacks. Data packets which meet defined criteria are detected and their details forwarded to a database server where the details are stored so as to be accessible for use in analysis in conjunction with the details of other detected packets. Packet detection uses a tap and a packet factory which creates a packet for analysis consisting of the received packet and a unique identifier. A series of adapters are used to apply functions to different parts of the packets, to detect those meeting the criteria.

A feedback-based scheme for improving TCP performance in ad hoc wireless networks

Abstract: Ad hoc networks are completely wireless networks of mobile hosts, in which the topology rapidly changes due to the movement of mobile hosts. This frequent topology change may lead to sudden packet losses and delays. Transport protocols like TCP, which have been designed for reliable fixed networks, misinterpret this packet loss as congestion and invoke congestion control, leading to unnecessary retransmissions and loss of throughput. To overcome this problem, a feedback scheme is proposed so that the source can distinguish between a route failure and network congestion. When a route is disrupted, the source is sent a route failure notification packet, allowing it to invalidate its timers and stop sending packets. When the route is reestablished, the source is informed through a route reestablishment notification packet, upon which it resumes packet transmissions. Simulation experiments show that in the event of route failures, as the route reestablishment time increases, the use of feedback provides significant improvements in performance.

Distributed control algorithms for service differentiation in wireless packet networks

Abstract: This paper investigates differentiated services in wireless packet networks using a fully distributed approach that supports service differentiation, radio monitoring and admission control. Service differentiation is based on the IEEE 802.11 distributed coordination function (DCF) originally designed to support best-effort data services. We extend the distributed coordination function to provide service differentiation for delay sensitive and best-effort traffic. Two distributed estimation algorithms are proposed and analyzed. A virtual MAC (VMAC) algorithm passively monitors the radio channel and estimates locally achievable service levels. The virtual MAC estimates key MAC level statistics related to service quality such as delay, delay variation, packet collision and packet loss. We show the efficiency of the virtual MAC algorithm and consider significantly overlapping cells and highly bursty traffic mixes. A virtual source (VS) algorithm utilizes the virtual MAC to estimate application level service quality. The virtual source allows application parameters to be tuned in response to dynamic channel conditions based on "virtual delay curves". We demonstrate through simulation that when these distributed virtual algorithms are applied to the admission control of the radio channel then a globally stable state can be maintained without the need for complex centralized radio resource management. Finally, we discuss a distributed service level management scheme that builds on the proposed algorithms to offer continuous service with handoff.

Edge adapter apparatus and method

Abstract: An apparatus and method for enhancing the infrastructure of a network such as the Internet is disclosed. A packet interceptor/processor apparatus is coupled with the network so as to be able to intercept and process packets flowing over the network. Further, the apparatus provides external connectivity to other devices that wish to intercept packets as well. The apparatus applies one or more rules to the intercepted packets which execute one or more functions on a dynamically specified portion of the packet and take one or more actions with the packets. The apparatus is capable of analyzing any portion of the packet including the header and payload. Actions include releasing the packet unmodified, deleting the packet, modifying the packet, logging/storing information about the packet or forwarding the packet to an external device for subsequent processing. Further, the rules may be dynamically modified by the external devices.

A Markov-based channel model algorithm for wireless networks

Abstract: Techniques for modeling and simulating channel conditions play an essential role in understanding network protocol and application behavior. In [11], we demonstrated that inaccurate modeling using a traditional analytical model yielded significant errors in error control protocol parameters choices. In this paper, we demonstrate that time-varying effects on wireless channels result in wireless traces which exhibit non-stationary behavior over small window sizes. We then present an algorithm that divides traces into stationary components in order to provide analytical channel models that, relative to traditional approaches, more accurately represent characteristics such as burstiness, statistical distribution of errors, and packet loss processes. Our algorithm also generates artificial traces with the same statistical characteristics as actual collected network traces. For validation, we develop a channel model for the circuit-switched data service in GSM and show that it: (1) more closely approximates GSM channel characteristics than a traditional Gilbert model and (2) generates artificial traces that closely match collected traces' statistics. Using these traces in a simulator environment enables future protocol and application testing under different controlled and repeatable conditions.

Modeling the effects of burst packet loss and recency on subjective voice quality

Abstract: A. INTRODUCTION Voice over IP networks differ from conventional telephone networks in that voice quality is affected by a wider variety of network impairments and can vary from call to call and even during a call. It is therefore desirable to monitor call quality in order that service providers can properly provision networks and that network resources are properly allocated. Passive monitoring systems examine operating characteristics of a system in order to assess or measure performance level. This may involve examining elements of the system, for example buffer levels, or examining the data stream being transmitted through the system. This contrasts with Active measurement systems in which test data is inserted into the system and used to obtain performance measurements. This paper describes a passive monitoring system (VQmon) for Voice over IP networks that is able to monitor per-call quality, providing feedback to a service management or CDR (Call Detail Record) system. The VQmon monitoring system also considers the effects of time varying impai rments such as bursty packet loss and recency. B. EMBEDDED PASSIVE MONITORING Passive monitoring systems examine operating characteristics of a system in order to assess or measure performance level. This may involve examining elements of the system, for example buffer levels, or examining the data stream being transmitted through the system. This contrasts with Active measurement systems in which test data is inserted into the system and used to obtain performance measurements. Embedded passive monitoring systems employ some form of monitoring function embedded into the equipment that comprises the system under test. This has the advantage of a closer relationship with system elements, allowing access to real time data and control information however has the disadvantage that implementation cost and complexity must be low. This contrasts with external passive monitoring systems which may, for example, be connected to T1 trunks or Ethernet LANs. Within the context of a VoIP network embedded passive monitoring can be integrated into VoIP Gateways, IP Phones or other end-systems, providing access on a per-call basis to CODEC selection, packet loss and delay information. This permits per-call estimates of transmission quality to be made with minimal impact on the service being monitored. Active monitoring systems typically make test calls through the VoIP network, transmit speech files and compare transmitted and received files using PSQM, PESQ or some similar method. This approach allows the CODEC performance to be directly measured however provides only …

Authenticating Streamed Data in the Presence of Random Packet Loss.

Abstract: We propose a new scheme for authenticating streamed data delivered in real-time over an insecure network The difficulty of signing live streams is twofold First, authentication must be efficient so the stream can be processed without delay Secondly, authentication must be possible even if some packets in the sequence are missing Streams of audio or video provide a good example They must be processed in real-time and are commonly exchanged over UDP, with no guarantee that every packet will be delivered Existing solutions to the problem of signing streams have been designed to resist worst-case packet loss In practice however, network loss is not malicious but occurs in patterns of consecutive packets known as bursts Based on this realistic model of network loss, we propose an authentication scheme for streams which achieves better performance as well as much lower communication overhead than existing solutions We have implemented our constructions as plug-ins to the RealSystem platform from Real Networks to authenticate audio and video streams

Dynamic behavior of slowly-responsive congestion control algorithms

Abstract: The recently developed notion of TCP-compatibility has led to a number of proposals for alternative congestion control algorithms whose long-term throughput as a function of a steady-state loss rate is similar to that of TCP. Motivated by the needs of some streaming and multicast applications, these algorithms seem poised to take the current TCP-dominated Internet to an Internet where many congestion control algorithms co-exist. An important characteristic of these alternative algorithms is that they are slowly-responsive, refraining from reacting as drastically as TCP to a single packet loss.However, the TCP-compatibility criteria explored so far in the literature considers only the static condition of a fixed loss rate. This paper investigates the behavior of slowly-responsive, TCP-compatible congestion control algorithms under more realistic dynamic network conditions, addressing the fundamental question of whether these algorithms are safe to deploy in the public Internet. We study persistent loss rates, long- and short-term fairness properties, bottleneck link utilization, and smoothness of transmission rates.

Method and system for packet communication employing path diversity

Abstract: Communication over lossy packet networks such as the Internet is hampered by limited bandwidth and packet loss. The present invention provides a path diversity transmission system for improving the quality of communication over a lossy packet network. The path diversity transmission system explicitly sends different subsets of packets over different paths, thereby enabling the end-to-end application to effectively see an average path behavior. Generally, seeing this average path behavior provides better performance than seeing the behavior of any individual random path. For example, the probability that all of the multiple paths are simultaneously congested is much less than the probability that a single path is congested. The resulting path diversity can provide a number of benefits, including enabling real-time multimedia communication and simplifying system design (e.g., error correction system design). Two exemplary architectures for achieving path diversity are described herein. The first architecture is based on source routing, and the second architecture is based on a relay infrastructure. The second architecture routes traffic through semi-intelligent nodes at strategic locations in the Internet, thereby providing a service of improved reliability while leveraging the infrastructure of the Internet.

Distributed video streaming over Internet

Abstract: With the explosive growth of video applications over the Internet, many approaches have been proposed to stream video effectively over packet switched, best-effort networks. A number of these use techniques from source and channel coding, or implement transport protocols, or modify system architectures in order to deal with delay, loss, and time-varying nature of the Internet. In this paper, we propose a framework for streaming video from multiple senders simultaneously to a single receiver. The main motivation in doing so is to exploit path diversity in order to achieve higher throughput, and to increase tolerance to packet loss and delay due to network congestion. In this framework, we propose a receiver-driven transport protocol to coordinate simultaneous transmissions of video from multiple senders. Our protocol employs two algorithms: rate allocation and packet partition. The rate allocation algorithm determines sending rate for each sender to minimize the packet loss, while the packet partition algorithm minimizes the probability of packets arriving late. Using NS and actual Internet experiments, we demonstrate the effectiveness of our proposed distributed transport protocol in terms of the overall packet loss rate, and compare its performance against a na*ve distributed protocol.

Method and apparatus for operating the internet protocol over a high-speed serial bus

Abstract: A method and apparatus of integrating the IEEE 1394 protocol with the IP protocol in which the IEEE 1394 high speed serial bus operates as the physical and link layer medium and the IP operates as the transport layer. There are differences in the protocols which require special consideration when integrating the two protocols. The IEEE 1394 configures packets with memory information and the IP operates under channel based I/O thereby necessitating a modification of the data transfer scheme to accomplish IP transfers over the IEEE 1394. Further, due to differences in packet headers, the IEEE 1394 packet header is modified to encapsulate IP packets. Moreover, in order to determine network packets quickly and efficiently, an identifier is inserted in each network packet header indicating that the packet should be processed by the network. Finally, in order to support the ability to insert or remove nodes on the network without a loss of data, the IP interface must not be disturbed. This is accomplished by maintaining constant IP addresses across bus resets which are caused by insertion or removal of nodes from the network.

Method and apparatus for allocation of resources

Abstract: A method and apparatus for allocating limited network resources, such as bandwidth and buffer memory, among various categories of data. Scheduler software adjusts the service weights associated with various data categories in order to regulate packet loss and delay. Central control software monitors network traffic conditions and regulates traffic at selected ingresses in order to reduce congestion at downstream bottlenecks. An advantageous method of calculating data utility functions enables utility maximization and/or fairness of resource allocation. Traffic at selected egresses is regulated in order to avoid wasting underutilized resources due to bottlenecks elsewhere in the network.

Joint source/FEC rate selection for quality-optimal MPEG-2 video delivery

Abstract: This paper deals with the optimal allocation of MPEG-2 encoding and media-independent forward error correction (FEC) rates under a total given bandwidth. The optimality is defined in terms of minimum perceptual distortion given a set of video and network parameters. We first derive the set of equations leading to the residual loss process parameters. That is, the packet loss ratio (PLR) and the average burst length after FEC decoding. We then show that the perceptual source distortion decreases exponentially with the increasing MPEG-2 source rate. We also demonstrate that the perceptual distortion due to data loss is directly proportional to the number of lost macroblocks, and therefore decreases with the amount of channel protection. Finally, we derive the global set of equations that lead to the optimal dynamic rate allocation. The optimal distribution is shown to outperform classical FEC scheme, thanks to its adaptivity to the scene complexity, the available bandwidth and to the network performance. Furthermore, our approach holds for any standard video compression algorithms (i.e., MPEG-x, H.26x).

Packet loss and delay performance of feedback and feed-forward arrayed-waveguide gratings-based optical packet switches with WDM inputs-outputs

Abstract: This paper analyzes the packet loss and delay performance of an arrayed-waveguide-grating-based (AWG) optical packet switch developed within the EPSRC-funded project WASPNET (wavelength switched packet network). Two node designs are proposed based on feedback and feed-forward strategies, using sharing among multiple wavelengths to assist in contention resolution. The feedback configuration allows packet priority routing at the expense of using a larger AWG. An analytical framework has been established to compute the packet loss probability and delay under Bernoulli traffic, justified by simulation. A packet loss probability of less than 10/sup -9/ was obtained with a buffer depth per wavelength of 10 for a switch size of 16 inputs-outputs, four wavelengths per input at a uniform Bernoulli traffic load of 0.8 per wavelength. The mean delay is less than 0.5 timeslots at the same buffer depth per wavelength.

FEC performance in multimedia streaming

Abstract: The performance of packet-level media-independent forward error correction (FEC) schemes are computed in terms of both packet loss ratio and average burst length of multimedia data after error recovery The set of equations leading to the analytical formulation of both parameters are first given for a renewal error process Finally, the FEC performance parameters are computed for a Gilbert (1960) model loss process and compared to various experimental data

Error control for receiver-driven layered multicast of audio and video

Abstract: We consider the problem of error control for receiver-driven layered multicast of audio and video over the Internet. The sender injects into the network multiple source layers and multiple channel coding (parity) layers, some of which are delayed relative to the source, Each receiver subscribes to the number of source layers and the number of parity layers that optimizes the receiver's quality for its available bandwidth and packet loss probability. We augment this layered FEC system with layered pseudo-ARQ. Although feedback is normally problematic in broadcast situations, ARQ can be simulated by having the receivers subscribe and unsubscribe to the delayed parity layers to receive missing information. This pseudo-ARQ scheme avoids an implosion of repeat requests at the sender and is scalable to an unlimited number of receivers, We show gains of 4-18 dB on channels with 20% loss over systems without error control and additional gains of 1-13 dB when FEC is augmented by pseudo-ARQ in a hybrid system, Optimal error control in the hybrid system is achieved by an optimal policy for a Markov decision process.

Network monitoring system with built-in monitoring data gathering

Abstract: The network monitoring system comprises a network router with built-in monitoring data gathering. The network router includes channels through which data pass in packets. Each of the packets includes a packet header. The network router additionally includes a header copier and a packet generator. The header copier generates a header copy from the packet header of at least some of the packets. The packet generator receives the header copies and forms monitoring data packets from them. Each monitoring data packet additionally represents temporal data relating to the header copies included in it. A method of obtaining performance data relating to a data transmission network that includes a node passes data through the node in packets. Each of the packets includes a packet header. At least some of the packet headers are copied to obtain respective header copies as monitoring data from which monitoring data packets are formed. The monitoring data packets additionally represent temporal data relating to the header copies included in them. The monitoring data packets are transmitted and the performance data are generated from the monitoring data contained in the monitoring data packets.

A bandwidth allocation/sharing/extension protocol for multimedia over IEEE 802.11 ad hoc wireless LANs

Abstract: We propose a novel bandwidth allocation/sharing/extension (DBASE) protocol to support both asynchronous traffic and multimedia traffic with the characteristics of variable bit rate (VBR) and constant bit rate (CBR) over IEEE 802.11 ad hoc wireless local area networks. The overall quality of service (QoS) will be guaranteed by DBASE. The designed DBASE protocol will reserve bandwidth for real-time stations based on a fair and efficient allocation. Besides, the proposed DBASE is still compliant with the IEEE 802.11 standard. The performance of DBASE is evaluated by analysis and simulations. Simulations show that the DBASE is able to provide almost 90% channel utilization and low packet loss due to delay expiry for real-time multimedia services.

Self-tuned congestion control for multiprocessor networks

Abstract: One-track performance in tightly-coupled multiprocessors typically, degrades rapidly beyond network saturation. Consequently, designers must keep a network below its saturation point by reducing the load on the network. Congestion control via source throttling-a common technique to reduce the network load-presents new packets from entering the network in the presence of congestion. Unfortunately, prior schemes to implement source throttling either lack vital global information about the network to make the correct decision (whether to throttle or not) or depend on specific network parameters, network topology or communication pattern. This paper presents a global-knowledge-based, self-tuned, congestion control technique that prevents saturation at high loads across different network configurations and commutation pattern. Our design is composed of two key components. First, we use global information about a network to obtain a timely estimate of network congestion. We compare this estimate to a threshold value to determine when to throttle packet injection. The second component is a self-tuning mechanism that automatically determines appropriate threshold values based on throughput feedback. A combination of these two techniques provides high performance under heavy load does not penalize performance under light load, and gracefully adapts to changes in communication patterns.

Reliable video communication over lossy packet networks using multiple state encoding and path diversity

Abstract: Video communication over lossy packet networks such as the Internet is hampered by limited bandwidth and packet loss. This paper presents a system for providing reliable video communication over these networks, where the system is composed of two subsystems: (1) multiple state video encoder/decoder and (2) a path diversity transmission system. Multiple state video coding combats the problem of error propagation at the decoder by coding the video into multiple independently decodable streams, each with its own prediction process and state. If one stream is lost the other streams can still be decoded to produce usable video, and furthermore, the correctly received streams provide bidirectional (previous and future) information that enables improved state recovery for the corrupted stream. This video coder is a form of multiple description coding (MDC), and its novelty lies in its use of information from the multiple streams to perform state recovery at the decoder. The path diversity transmission system explicitly sends different subsets of packets over different paths, as opposed to the default scenarios where the packets proceed along a single path, thereby enabling the end-to-end video application to effectively see an average path behavior. We refer to this as path diversity. Generally, seeing this average path behavior provides better performance than seeing the behavior of any individual random path. For example, the probability that all of the multiple paths are simultaneously congested is much less than the probability that a single path is congested. The resulting path diversity provides the multiple state video decoder with an appropriate virtual channel to assist in recovering from lost packets, and can also simplify system design, e.g. FEC design. We propose two architectures for achieving path diversity, and examine the effectiveness of path diversity in communicating video over a lossy packet network.

Data packet numbering in packet-switched data transmission

Abstract: A method and a telecommunication system for data packet numbering in packet-switched data transmission in connection with a handover, in which the responsibility for a connection is transferred from the connection between a mobile station and a first wireless telecommunication network to the connection between said mobile station and a second wireless telecommunication network. In the first wireless telecommunication network a data packet number space available for data packet numbering is bigger than a data packet number space of the second wireless telecommunication network. Data packet numbering is restricted in the first wireless telecommunication network such that the numbers of the data packets of the first wireless telecommunication network do not exceed the maximum value of the data packet number space of the second wireless telecommunication network.

Buffer overflow management in QoS switches

Abstract: We consider two types of buffering policies that are used in network switches supporting QoS (Quality of Service). In the FIFO type, packets must be released in the order they arrive; the difficulty in this case is the limited buffer space. In the bounded-delay type, each packet has a maximum delay time by which it must be released, or otherwise it is lost. We study the cases where the incoming streams overload the buffers, resulting in packet loss. In our model, each packet has an intrinsic value; the goal is to maximize the total value of packets transmittedOur main contribution is a thorough investigation of the natural greedy algorithms in various models. For the FIFO model we prove tight bounds on the competitive ratio of the greedy algorithm that discards the packets with the lowest value. We also prove that the greedy algorithm that drops the earliest packets among all low-value packets is the best greedy algorithm. This algorithm can be as much as 1.5 times better than the standard tail-drop policy, that drops the latest packets.In the bounded delay model we show that the competitive ratio of any online algorithm for a uniform bounded delay buffer is bounded away from 1, independent of the delay size. We analyze the greedy algorithm in the general case and in three special cases: delay bound 2; link bandwidth 1; and only two possible packet values.Finally, we consider the off-line scenario. We give efficient optimal algorithms and study the relation between the bounded-delay and FIFO models in this case.

Performance enhancement of transmission control protocol (tcp) for wireless network applications

Abstract: A new Fast Recovery Plus (FR+) mechanism, and associated method, for wireless and/or mobile network applications to avoid network congestion in a TCP/IP packet-switched network. A method of flow control and congestion avoidance congestion in a network determining, at the source node, if packet loss is due to transmission error; and if the packet loss is due to the transmission error, setting, at the source node, the slow start threshold Ssthresh to Max (CWND, (Ssthresh + CWND)/2), wherein CWND and Ssthresh exhibit values recorded, when the packet lost was detected.

Video communication method and system employing multiple state encoding and path diversity

Abstract: Video communication over lossy packet networks such as the Internet is hampered by limited bandwidth and packet loss. The present invention provides a system for providing reliable video communication over these networks, where the system includes at least two jointly designed subsystems: (1) multiple state video coding system and (2) path diversity transmission system. Multiple state video coding combats the problem of error propagation that results from packet loss by coding the video into multiple independently decodable streams, each with its own prediction process and state. If one stream is lost the other streams can still be decoded to produce usable video, and furthermore, the correctly received streams provide bidirectional (i.e., previous and future) information that enables improved state recovery for the corrupted stream. The path diversity transmission system explicitly sends different subsets of packets over different paths, as opposed to the prior art approaches where the packets proceed along a single path. By explicitly sending different subsets of packets over different paths, the path diversity transmission system enables the end-to-end video application to effectively see an average path behavior, which is referred to herein as path diversity. Generally, seeing this average path behavior provides better performance than seeing the behavior of any individual random path. The resulting path diversity provides the multiple state video decoder with an appropriate virtual channel to assist in recovering from lost packets, and can also simplify system design (e.g., forward error correction design).

Controlling network congestion using a biased packet discard policy for congestion control and encoded session packets: methods, systems, and program products

Abstract: A packet network congestion control system using a biased packet discard policy includes a plurality of end points having codecs operating in a framework, e.g. ITU-T H.323 protocol to establish a communication session. The protocol enables the codecs to negotiate codec type and associated parameters. Once a connection and session are established, compressed voice and data packets start flowing between the two end points. A control entity supplies congestion control packets periodically. The control packets provide a “heartbeat” signal to the codec at the other end of the session. Each codec receiver uses the “heartbeat” signal as an indication of network congestion. As network congestion increases, routers within the network discard excess packets to prevent network failure. The network discards all packets classified as congestion control packets whenever a flow control mechanism detects congestion or a trend toward congestion. As packets are discarded, the end points renegotiate codec type and/or parameters to realize lower bit rates.

Codec with network congestion detection and automatic fallback: methods, systems & program products

Abstract: A codec detects congestion in a packet network and responds via a session control protocol to re-negotiate codec-type and/or parameters with the receiving codec to reduce bit rate for supporting a session. Once the connection and session are established, encoded packets start flowing between the two codecs. A control entity sends and receives network congestion control packets periodically in the session. The congestion control packets provide a “heartbeat” signal to the receiving codec. When the network is not congested, all “heartbeat” packets will be passed through the network As network congestion increases, routers within the network discard excess packets to prevent network failure. The codecs respond to the missing packets by slowing down the bit rate or proceeding to renegotiate a lower bit rate via the session control protocol. If there are no missing packets, the codecs detect if the session is operating at the highest bit rate, and if not, re-negotiate a higher bit rate.