Most data networks are packet-switched. Data is communicated over a packet-switched network in small chunks, or "packets", which require no dedicated circuit. Each packet contains information that allows the data network to route it to the appropriate destination. Packets from many different senders travel sequentially over single connections between routing points, and packets from the same sender may travel different routes as network conditions change. Consequently, consecutive packets from a specific sender to a specific receiver may experience different delays as they travel different routes or experience different competing traffic loads along the network.
Researchers have sought ways to communicate real-time information over packet-switched data networks in order to take advantage of the time-varying nature and information redundancies found in most real-time data. For example, it is now possible to route voice telephone traffic over data networks through a technique commonly referred to as "Voice Over IP", or "VoIP" for short. VoIP can require significantly less average bandwidth than a traditional circuit-switched connection for several reasons. First, by detecting when voice activity is present, VoIP can choose to send little or no data when a speaker on one end of a conversation is silent, whereas a conventional, circuit-switched telephone connection continues to transmit during periods of silence. Second, the digital audio bitstream utilized by VoIP may be significantly compressed before transmission using a codec (compression/decompression) scheme. Using current technology, a telephone conversation that would require two 64 kbps (one each way) channels over a circuit-switched network may utilize a data rate of roughly 8 kbps with VoIP.
The variation in packet arrival rate, or "jitter", existing on most packet networks, presents challenges for real-time communication. To compensate for jitter, a real-time receiver must buffer packets for an amount of time sufficient to allow orderly, regular playout of the packets. Researchers have long recognized the need for an accurate method of receiver playout buffer length selection in real-time packet data communications such as VoIP. If the buffer delay is too short, "slower" packets will not arrive before their designated playout time and playout quality suffers. If the buffer delay is too long, it noticeably disrupts interactive communications. Selection of a near-optimal packet buffer delay for real-time communications requires accurate knowledge of actual packet delays.
Various protocols have been suggested for allowing receivers to obtain delay information. These include two described by W. Montgomery, "Techniques for Packet Voice Synchronization", IEEE J. on Selected Areas in Comm., vol. SAC-1, No. 6, pp. 1022-1028, Dec. 1983. One protocol uses an absolute clock reference by both a sender and a receiver. The sender timestamps each packet, and the receiver compares the timestamps on packets it receives to the absolute clock reference to determine delay. A second protocol would require that each packet switch along the network update a packet delay field to include the amount of time the packet was delayed by the switch. Since switches are the major source of variations in delay, the receiver can estimate delay by examining the delay field in received packets.
Unfortunately, neither of the protocols mentioned above are in widespread use today. Instead, most real-time packet data transmissions utilize the Real-time Transport Protocol (RTP). A sender using this protocol includes a packet timestamp generated from a local clock. The clock rate used to generate consecutive RTP timestamps is the clock rate of the data being transmitted--thus two consecutive packets should carry timestamps that differ by the number of data samples contained in the first of the two packets. Although RTP timestamps allow a receiver to reassemble samples in correct order, they contain no absolute delay information because the sender and receiver local clocks are not synchronized.
Despite the lack of absolute delay information in RTP headers, researchers have found ways to use adaptive, rather than fixed, buffer delays with RTP data streams. Although a fixed playout buffer delay can work in some circumstances (particularly with real-time communication over local area networks), adaptive playout buffer delay methods will generally perform better over a range of network conditions. An adaptive method attempts to minimize delay for current network conditions. Most techniques for adaptively adjusting buffer delay base their adjustments on statistics gleaned from RTP (or similar) timestamp histories. Four such techniques are discussed in R. Ramjee, et al., "Adaptive Playout Mechanisms for Packetized Audio Applications in Wide-Area Networks" in Proceedings of the Conference on Computer Communications (IEEE Infocom), (Toronto, Canada), pp. 680-688, June 1994.
Each technique discussed in Ramjee et al. computes a delay estimate d.sub.i and a delay variation v.sub.i for each packet i. The basic adaptive algorithm is illustrated in FIG. 1. A packet i, containing a timestamp ts.sub.i affixed to packet i by the sender, is received from packet-switched network 20 by receiver 16. Summer 24 subtracts timestamp ts.sub.i from a receive timestamp tr.sub.i, taken from receiver clock reference 22, to produce a difference sample n.sub.i. With RTP, this difference will include an offset equal to the difference between the sender and receiver clock references. First-order filter 26 computes a mean delay estimate d.sub.i from difference samples n.sub.i. Summer 28 feeds the absolute value of the difference between d.sub.i and n.sub.i to a second filter 30, which uses these samples to create a filtered estimate of the variation in delay v.sub.i. Multiplier 32 produces a multiple k of v.sub.i, which summer 34 adds to d.sub.i and ts.sub.i to create a playout time p.sub.i for packet i.
Ramjee et al.'s other three discussed methods comprise various heuristic adaptations of the adaptive playout delay estimator of FIG. 1. One adaptation uses different time constants for filter 26, depending on whether the latest measurement n.sub.i will increase or decrease delay estimate d.sub.i. Another adaptation suspends delay estimate filtering temporarily if it detects a "spike" in the packet arrival rate. A fourth algorithm dispenses with filter 26 altogether, by examining all n.sub.i computed for the last talkspurt received and setting d.sub.i to the minimum of these values for the next talkspurt.
USPTO Backgrounds
USPTO Backgrounds