Review of Evaluation

Quality of Real-time video transmission service is crucial to the function of multimedia providers.
The research here involved using NetEm emulator to serve the purpose of a router; which
functions with VMWare to adjust settings for Bandwidth, Network Adaptors and Packet losses.
Two computers were connected to communicate over a video call and share packet data through
the router provided. The video call simulation software is Riverbed Modeler Academic Edition
version 17.5. It was used to select the application configuration for interactive multimedia, profile
definitions, statistics and packet delay variations.

The results from our in-depth research reflected how jitters being Packet losses can have notable
influence through IP networks on video communication. It however does not cause the network
conn problems that we assumed during the research; it rather affected the quality of images
during the call transmission. The jitters and losses in Packet data dropped the quality of
interactive video and audio transmissions because of packet traffic drop. This Traffic drops mean
that the number of data packets moving across IP interfaces reduced at all the nodes in the
network.

Below are the different drops in video quality results observed due the project:

• Frame freeze:

An image is steady on display for a while on the screen; which is caused by loosing continuous
MPEG frame, while the configuration tries to restore the I-frame flow.

• Slicing:

This occurs with packet drops in I-frame, P-frame or B-frame when on video playback; it is
capable of extending to the GOP if the error affects a corresponding frame. The video quality will
drop, but it can be corrected when a decoder receives a good I-frame to clear up the picture.

• Ghosting:

This is caused by a missing bulk if I-frame through a packet loss around the scene shift. The
decoder can repair this error when it receives an I-frame also.

• Blocking:

On this observation, the I-frame or P-frame gets broken by loss of packet data and the
subsequent frames do not have references from the decoder. The impairment is obvious on
videos with heavy action. The error is corrected when the lagging I-frame is received by the
decoder.

These results were achieved under the thorough research conditions with the best simulation
configuration for video calling. Nonetheless, we had the following professional difficulties as we
moved forward; we understand that the NetEm software may project a lower function since it is
an emulator, the assurance for network topography is minimal and the people tend to be
unavailable at crucial situations.

Dealing with a high motion video application over a packet data network can result in increased
error on the video quality from losses of dynamic pictures on the IP network. This confirms the
importance of including regulations on Delays, jitters and Packet losses in Service Level
Agreements.

Conclusion and Addition

With the current level of our communication technology, a wide range of multimedia data can be
transferred over IP networks. Our focus on this project is the video data which involves large data
packets; the traffic from this transfer results in a complicated network with the problems of delays,
packet loss and jitters.

The research succeeded with discovering and understanding the problem of packet delay and
losses over voice calls through IP network. We searched to find possible solutions to the errors
which landed at the proposed method by IETF engineers which include the following;

• RFC 3550: A Transport Protocol for Real-Time Applications;

• RFC 3551: RTP Profile for Audio and Video Conferences with Minimal Control; are the
contained real time technologies

Other possible methods which can help solve the problems which this study exposes involves;
using De-Jitter buffer which eliminates the delay variables. There is also an admission control
method that measures packet transfer; these two alternatives have their significant roles in total
delay rectification.

With the earlier concept which the internet was introduced with, it was designed to display steady
images of webpage which do not need immediate transmission of real time packet data. This
posed more challenges with the introduction of VoIP technology, and more improvements have
been seen since then. More study is being carried out in Asia to improve video and
telecommunications over the internet using interactive video audio systems involving
communication servers and big data analysis.

These methods assure users of a better internet video communication without so much
disturbance in transmission. The future of the video audio telecommunications technology is
actually seeing promising improvements.