Paper review : Adaptive Playout Mechanisms for Packetized Audio Applications in
Wide-Area Networks (RKTS94)
Reviewer : Hai Fang (hfang@acm.org)
- Goal
To eliminate the end to end delay jitter for audio data transmission over the
Internet.
- Contribution
This paper proposed an adaptive playout buffer mechanism used to reduce network
jutter effects on audio information, based on the tradeoff between the toleratable
delay and the packet loss.
- Main ideas
- Given strict end to end delay and interframe delay requirements for real time
transmissions, packets delayed over a certain time limit can be considered lost
(lies outside the buffer).
- A receiver-based algorithm can adapt fast to the possible spikes by checking
the delay between consecutive packets.
- Evaluation
- Significance rating: 3
This paper observes the frequently occurenced spikes in end-to-end audio
packets transmission, hence presents a refined algorithm which take this factor
into the delay estimation.
- Convincing rating
Simulations and experiments on Internet are used in this paper to get a performance
comparison. Th authors also explain the key idea, which is about the spikes and
analyzed the mathematical model of it.
- Limitation
The term adaptive means refinement based on the existed algorithms, however,
there are some restrictions for this kind of algorithms: they are receiver-based,
using the stochastic estimation of the packet delay, and tradeoff tradeoff the
the delay (buffer size) and the packet loss.
- Conclusion
Although stochastic method is a good way to set the buffer size, the packet burst
on the real Internet still need to take care of and deserves a special refinement.
10/22/01