Simulation/Implementation of enhancement algorithms for VOIP
$30-250 USD
Suljettu
Julkaistu noin 9 vuotta sitten
$30-250 USD
Maksettu toimituksen yhteydessä
Problem Statement
Synchronization in IP telephony in Enterprise Networks by simulating the existing jitter buffer algorithms and them enhancing them.
Components of the project -
A] VOIP traffic generation - This should comprise of voice and video data packet generation/simulation
B] Scaling up of the network to use large number of nodes (100 or more nodes)
C] Implementation of fixed jitter buffer algorithm
D] Implementation adaptive jitter buffer algorithm
The simulation should also include use of codecs (combination of coder and decoder) in it
The most popular voice coding ITU-T for telephony and packet voice include G.711, G.726, G.728 and G.723.1
The implementation can include any of these.
Since this is research work, new work/enhancements need to be implemented
Hence it is Important to enhance these algorithms and compare the results before and after algorithm enhancements.
Ehancement to the above mentioned synchronization algorithms is to be carried out by modifying the bit rate of the codecs in order minimize packet loss while maintaining the output quality
Input details -
a) VOIP traffic (Comprising of voice and video data packets) should be generated. This VOIP traffic should be subjected to the Enterprise network simulation
Output Details -
Output should be captured under two cases i.e. with and without application of synchronization algorithms. The details are as follows -
a) Output VOIP data stream (voice and video ) playout after passing through the Enterprise Network without application of any of the synchronization algorithms
b) Output VOIP data stream (voice and video ) playout after passing through the Enterprise Network and after getting subjected to the synchronization algorithms
Metrics/Measurements -
The following metrics/measurements should be captured -
a) End to end delay (Latency) in milliseconds
b) Jitter (Variation in delay) in milliseconds
c) Packet loss (%)
d) Inter-stream latency
- This refers to the relative latencies that can be encountered between the audio and video data streams and is based on how the relative average transit time for the given streams, at any given point, vary from each other
These metrics would be compared before and after application of synchronization algorithm so as to arrive at the conclusion that the desired synchronization in VOIP traffic in Enterprise Networks has been achieved
Would need phase-wise delivarables of the project.
Need the VOIP traffic generation part immediately.
Thanks and Regards,
Gary