Paper review : Adaptive Playout Mechanisms for Packetized Audio Applications in
Wide-Area Networks (RKTS94)
Reviewer : Hai Fang (email@example.com)
To eliminate the end to end delay jitter for audio data transmission over the
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.
- 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.
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.
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.