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Following the paper [Ramjee 1994], we now describe a generic algorithm that the receiver can use to adaptively adjust its playout delays To this end, let ti = timestamp of the ith packet = the time packet was generated by sender ri = the time packet i is received by receiver pi = the time packet i is played at receiver The end-to-end network delay of the ith packet is ri - ti Due to network jitter, this delay will vary from packet to packet Let di denote an estimate of the average network delay upon reception of the ith packet This estimate is constructed from the timestamps as follows: di = (1-u) di-1 + u (ri - ti) where u is a fixed constant (eg, u = 01) Thus di is a smoothed average of the observed network delays r1 - t1,, ri - ti; the estimate places more weight on the recently observed network delays than on the observed network delays of the distant past This form of estimate should not be completely unfamiliar; a similar idea is used to estimate round-trip times, as discussed in 3 Let vi denote an estimate of the average deviation of the delay from the estimated average delay This estimate is also constructed from the timestamps: vi = (1-u) vi-1 + u | ri - ti - di | The estimates di and vi are calculated for every packet received, although they are only used to determine the playout point for the first packet in any talkspurt Once having calculated these estimates, the receiver employs the following algorithm for the playout of packets If packet i is the first packet of a talkspurt, its playout time, pi, is computed as: pi = ti + di + Kvi, where K is a positive constant (eg, K = 4) The purpose of the K vi term is to set the playout time far enough into the future so that only a small fraction of the arriving packets in the talk spurt will be lost due to late arrivals The playout point for any subsequent packet in a talkspurt is computed as an offset from the point in time when the first packet in the talkspurt was played out In particular, let qi = pi -ti , be length of time from when the first packet in the talk spurt is generated until it is played out If packet j also belongs to this talk spurt, it is played out at time pj = tj + qi The algorithm just described makes perfect sense assuming that the receiver can tell whether a packet is the first packet in the talk spurt If there is no packet loss, then the receiver can determine whether packet i is the first packet of the talk spurt by comparing the timestamp of the ith packet with the timestamp of the (i-1)st packet Indeed, if ti - ti-1 > 20 msec, then the receiver knows that ith packet starts a new talkspurt But now suppose there is occassional packet loss In this case two successive packets received at the destination may have timestamps that differ by more than 20 msec when the two packets belong to the same talkspurt So here is where the sequence numbers become useful The receiver can use the sequence numbers to determine whether the > 20 msec difference in timestamps is due to a new talkspurt or to lost packets.

Another concern with log replay is that the database on the remote side is not, in fact, identical to the one on the primary side As a result, it would be more difficult, and, in fact, probably impossible, to take an extract from it and apply that extract back to the primary side, in the event of data corruption As we mentioned previously, timestamps and record identifiers are going to be different on the remote side If it were necessary to copy the replicated database back to the primary database server, serious confusion and corruption will result However, database log replay is a relatively big win because it requires no application changes to implement replication The database administrators can configure the log copy and recovery steps, setting the time interval between copies and the verification of updates on the replicated machines.

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We have discussed in some detail how an Internet phone application can deal with packet jitter We now briefly describe a few schemes that attempt to preserve acceptable audio quality in the presence of packet loss Such schemes are called loss recovery schemes Here we define packet loss in a broad sense: a packet is lost if either it never arrives at the receiver or if it arrives after its scheduled playout time Our Internet phone example will again serve as a context for describing the loss recovery schemes As mentioned at the beginning of this section, retransmitting lost packets is not appropriate in an interactive real-time application such as Internet phone Indeed, retransmitting a packet that missed its playout deadline serves absolutely no purpose And retransmitting a packet that overflowed a router queue can not normally be accomplished quickly enough Because retransmissions are inappropriate, Internet phone applications often use some type of loss anticipation scheme Two types of loss-anticipiation schemes are forward-error correction (FEC) and interleaving Forward-Error Correction (FEC)

Some applications are better suited to this technique than others The following is a brief checklist of features that allow an application to fit the log replay model:.

) CADs are also present at merchant sites (usually in the form of point-of-sale devices) and at issuing banks These CADs may look much like a typical point-of-sale credit card processor, with a small display screen and a few buttons for answering queries or entering PINs In similar fashion to today's ATMs, a smart card is inserted directly into the reader The reader then provides power to the card and sets it up to receive software commands Although a CAD of this sort may look like an ATM on the outside, a smart card is much more powerful than an ATM card or a mag-stripe credit card These latter cards store only a few kilobytes of information on the magnetic stripe and are not capable of performing computations Smart cards can store many megabytes of information and carry out sophisticated cryptographic calculations.

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Applications that can easily detect missing transactions when connecting to a database for the first time and that know how to find the right starting point. This is a critical consideration for stating that no changes are necessary to the application. If the application can look for transactions that fell into the gutter, then it will tolerate log replication well. Low volumes of inserts and updates so that the log files do not grow rapidly. Applications with high rates of change generate large log files and can induce a backlog on the replica database servers as they struggle to keep up. Applications that can tolerate a small time skew between the live and redundant sites. If the primary site goes down and the application has to reconnect to the redundant site, it may have to wait for several transactions to be played through the recovery manager before it can reconnect to the database. This delay is likely to be similar to the delay imposed by a failover.

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