Broadband Multimedia Networks Assignment

This assignment discusses various topics related to broadband multimedia networks, including MPEG-DASH server, MPD (Media Presentation), bit rates, queuing algorithms, RTP-based VoIP calls, SIP client and server, and QoS for UNI-V traffic. It provides insights into the concepts and technologies used in broadband multimedia networks. Subject: Broadband Multimedia Networks, Course Code: N/A, Course Name: N/A, College/University: N/A

Broadband Multimedia Networks Assignment

Question 1
In MPEG-DASH server movie is stored as multiple files (Kapetanakis, Zampoglou,
Malamos, Panagiotakis & Maravelakis, 2014). Each file is segmented as different quality levels.
Because MPEG-DASH player facilitates the adaptive bit rate feature. This feature helps to select
the highest possible bitrate based on the current network strength etc. So that only the movie is
stored at different bit rates.
MPD stands for Media Presentation Description. It describes the segment information
like timing, resolution, etc. Here one or more representation of the video files are there in the
server. Here the bit rate is selected based on the network conditions and user preferences
(Kesselman, Kogan & Segal, 2010).
MPEG-DASH Server
One of the well sought and applicable bitrate streaming processes in the market which enables
the streaming of the media in high quality by the internet is MPEG-DASH; service delivered by
HTTP Servers. This is one of the first streaming processes that are of international standard and
works on adaptive bitrate HTTP. This server is sometimes confused with a transport protocol,
whereas the protocol used by this server is TCP (Kapetanakis et al., 2014) (Park and Kim, 2015)
(Thomas et al., 2017).
MPD (Media Presentation)
Segment information such as Media properties, URL, and Timings are described by a Media
Presentation Description Commonly termed as MPD. This MPD can be classified in a number of
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ways according to the specification of the segment such as base, template, list, timeline and use
case. Segment are the parts which can bear any kind of data related to media, although the
certain set of instructions are to be used for two types of containers, A Mp4 file format which
lies on the ISO file as a base and MPEG, a type of system which uses 2 different transport
streams.
Question 2
200 kbps needs 1 sec. 400 kbps needs 2 seconds. This is for bit rate 200kbps
500 kbps needs 1 sec. 1000 kbps needs 2 seconds. This is for bit rate 500kbps
1000 kbps needs 1 sec. 2000 kbps needs 2 seconds. This is for bit rate 1000kbps
200 kb will be produced in 1 sec. 1600 kb will be produced in 8 seconds. This is for 200-bit rate.
500 kb will be produced in 1 sec. 1600 kb will be produced in 8 seconds. This is for500-bit rate.
1000 kb will be produced in 1 sec. 8000 kb will be produced in 8 seconds. This is for 1000 bit rate
seconds (Mustafa & Talab, 2016).
Time Requirement
Chunk size 400Kb 400/1600 0.25 sec
Chunk size 1000Kb 1000/1600 0.625 sec
Chunk size 2000Kb 2000/1600 1.25 sec
Chunk size 1600Kb 1600/1600 1 sec
Chunk size 4000Kb 4000/1600 2.5 sec
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Chunk size 8000Kb 8000/1600 5 sec
The semantic gap is one of the toughest factors while creating segmentation for any given movie
towards a semantically correlated units. This does not feed you with information necessary to
correlate between the different shot and sometimes it is even harder to detect the scenes which
bear the capability of dynamic content. In this particular article, the method proposed is the
representation of key frames of the video shots by the local descriptors that do not vary. The
method proposed in this particular article is proved by the numerical analysis and states this
method results in higher detection rates. The analysis also opines that the given method will
highly likely to make a good balance between passion and recall.
This method proposed is temporal soothing that is carried out upon words histogram that is
extracted from the video shots. For every under consideration shot, any particular number of key
frames may be extracted and then these key frames are assessed upon the scale of local
descriptors that are invariant. Every possible descriptor may be joined and form a cluster and
then made a visual word of it. For every given word a histogram of visual words is computed
accordingly. This histogram later than managed according to their preceding or exceeding visual
words using a Gaussian kernel which helps in the preservation of the information effectively. By
adjusting these by the Gaussian kernel the detection of both scene and boundaries of any under
study video.
Question 4
 The tail-drop is the simplest form of queuing algorithm, used at the time of writing in
internet routers. It provides the responsibility for resource allocation and congestion
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control in the network. For responding and detecting the congestion, TCP has complete
responsibility.
 The buffer bloat is the main reason for the latency between the packet switching network.
It was done because of overloading the packet buffers. It will also create variation in
packet delay, at the same time it reduces the throughput for the overall network.
 FIFO queuing
The concept of queuing, the one that is also known for like the first come, and first served
queuing is quite simple. Always the data packet that is transmitted from the router will be the
first one that was received. The illustration is depicted in Figure 1(a), the figure defining the
FIFO that illustrates with “slots” this carries packets of data eight to be precise. While it is
known that the buffer space of any given router is finite, and there is a data packet that
arrives and has to queue due to that space already utilized, then such packet will be discarded
by the router. This particular phenomenon is termed as tail drop, because of the packets
arriving at the tail of the router are discarded. The FIFO and tail drop are two different terms
(Mustafa and Talab, 2016). FIFO is more of schedule planner, which organizes the order of the
packets and how are these packets to be submitted, whereas the tail drop is a discarding
system, which defines and outlined which packets are to be dropped. As a matter of fact, the
FFO and tail drop, both of these are the simplest possible ways for queue management and
discarding packets policy. Sometimes both of these are considered collectively and are
termed as Vanilla bundle for implementation. Unluckily, this is commonly termed as FIFO
queuing while is should possibly be termed as FIFO with a tail drop. This is the simplest of
the algorithms, available for queuing and is extensively used by routers on the matters of
writing. The simplest possible way of queuing results in the transfer of all the responsibility
regarding the proper allocation of resources and control of congestion to the peripherals
within a network (Kesselman, Kogan and Segal, 2010) (Ariestya, Praptiningsih, and Saputra,
2015).
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Question 5
Downward speed is 12 Mbit/sec. 12582912 bits in a second.
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Upward speed is 1 Mbit/sec. 1048576 bits in a second.
24 two way RTP based voice calls will be made.
RTP based VoIP calls
A standard format of packing which delivers the audio and the video by the help of real-time
transport protocol (RTP) over the internet. While the commissioning of the specific connections
within a network, This RTP provides a foundation with the integration of VoIP, This system
works with the conjunction made with SIP (Alajmi, Haj Aliwi, and Alieyan, 2017).
The voice stream of audio is transmitted or received by the same protocol RTP. This protocol is
responsible for carrying the payload in terms of voice from the point of transmittance to the point
where it should be received. Payload should be sought as steam of continuous packets that run
parallel along with the network.
Most of the communications carry two streams, delivering one at every endpoint (Adeyeye &
Ventura, 2010). If this is to be sought as a two-way process, at any given instant every remote
endpoint is continuously transmitting and receiving packets. The call quality issues that arise
within the system of-of VoIP can be cut down if the right packet is carried in terms of the
performance of the network.
202.1.84.1 is the SIP Proxy. 202.1.84.20 is the client. 100 hops. 100 hops.
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SIP Client
A real-time communication can be established by the use of SIP Client. As for this, the
softphone functionality often lies within SIP and other possible features (Alajmi, Haj Aliwi &
Alieyan, 2017). Say, SIP client can provide its customer with a video, file transfer, chat and can
even grant access to the remote desktop. Most of the SIP Clients are of open source nature, an
average user can utilize these for free while some of it is offered against money. Other may still
be excluded up to an extent of any particular provider, like VoIP provider. What brings this
much functionality to the SIP that it is not even tied to any service in particular. Major clients of
SIP are SIP Droid, Bria, and 3CX, and a number of business providers recommend these. A SIP
client can be of any nature ranging from mobile phone to desktop, for the most developers this is
kept in mind and they release for both (Adeyeye and Ventura, 2010).
SIP Server
A SIP server is the principle part of an IP PBX and essentially manages the administration of all
SIP brings in the system. A SIP server is additionally alluded to as a SIP Proxy or a Registrar.
The SIP server does not actually transmit or get any media – this is finished by the media server
in utilizing the RTP convention. Inside the setting of an IP-PBX condition, it is quite often
obvious that the SIP server and its Media server buddy dwell on a similar machine (Yavas,
Hokelek, and Gunsel, 2017).
SIP servers acknowledge SIP demands and react to them. A SIP server is an application that may
follow up for the benefit of a SIP customer or client specialist (UA) or may give data or bearing
to a UA. There are a few kinds of SIP servers including intermediary, divert, and enlistment
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19 phone calls. 15880 SIP control packets. 60500 RDP pockets. 10 inward phone call and 9
outward phone calls are made.
SIP control packets
Session Initiation Protocol (SIP) is a standout amongst the most well-known protocols utilized
in VoIP innovation. It is an application layer protocol that works related to other application
layer protocols to control interactive media correspondence sessions over the Internet. SIP is a
signaling protocol used to make, alter, and end sight and sound session over the Internet
Protocol. A session is only a straightforward call between two endpoints. An endpoint can be a
cell phone, a workstation, or any gadget that can get and send interactive media content over the
Internet (Goncalves & Iancu, n.d.).
RTP packets
RTP parcels are made at the application layer and gave to the vehicle layer for conveyance.
Every unit of RTP media information made by an application starts with the RTP parcel header
(Jung, 2013).
The RTP header has a base size of 12 bytes. After the header, discretionary header
augmentations might be available. This is trailed by the RTP payload, the organization of which
is controlled by the specific class of application. [20] The fields in the header are as per the
following:
Time Stamp, Header Extension, CSRC, SSRC, Version, P (padding), X (extension), M (Marker),
PT (payload type),sequence number (Held, 2016).
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Question 8
4.20 4.40 4.60 4.80 5.00 5.20 5.40
0
20
40
60
80
100
120
140
160
180
Histogram of "some data"
Time (ms)
Frequency
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4.20 4.40 4.60 4.80 5.00 5.20 5.40
0
10
20
30
40
50
60
70
80
90
100
CDF of "some data"
Time (ms)
%
RTP Frames
Question 9
 Opened the trace file
 1st and 2nd APR packets represent the ARP request sent by the computer
 ARP Reply sent to the computer with the ARP-requested Ethernet address.
Question 10
 UNI-V uses “SIP” signaling protocol CPE and access provider.
 UNI_V uses G.711 A law commanding codec
 Packetisation rate for codec is 20ms.
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 Methods used by NBNCo to ensure the QoS for UNI-V traffic traversing
o TC_1 traffic class
o SIP signaling with RTP stream
o G.711 A law commanding codec
Any peripheral equipment of the server system which is used at the end of any particular
communication server system which is physically installed on the premises of subscriber of the
server. Such equipment is the type of equipment which are often connected with the complex
circuits of the server. There are some of the equipment such as routers, telephones, and modems
which are mostly considered as CSE. Such equipment basically works as the gateway for the end
to end communication within a server to provide with the data necessary for the customer.
CPE equipment can often be characterized into generally two of the possible terms, active and
passive. The routers and residential gateways fall under the category of active CSE Equipment,
whereas the telephones can be classified as the passive type of CSE Equipment. CSE Can
sometimes be also referred to as the equipment which is bought by the customer or the
equipment that is sold separately by the service provider. There is also a point to understand that
not all the equipment that is utilized in the communication server can generally be classified as
the CSE equipment. Some of the equipment that is less directly related to the server such as
Manuals, Network Cables and Adapter, thus such equipment are sub-classified or termed as CSE
peripherals. As the technology is under evolution exponentially with the time, CSE equipment is
more modernized towards hybrid equipment. The hybrid equipment generally bears the
capability to carry out the data packets both in analog as well as digital signals.
References
Adeyeye, M., & Ventura, N. (2010). A SIP-based web client for HTTP session
mobility and multimedia services. Computer Communications, 33(8), 954-964.
doi: 10.1016/j.comcom.2010.01.015
Alajmi, N., Haj Aliwi, H., & Alieyan, K. (2017). VoIP Protocols' Bandwidth Based-
Mini/RTP Header Using Different Codecs: A Comparison. Asian Journal Of
Scientific Research, 10(3), 110-115. doi: 10.3923/ajsr.2017.110.115
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Belloc, H. (1967). On. Freeport, N.Y.: Books for Libraries Press.
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Goncalves, F., & Iancu, B. Building telephony systems with OpenSIPS.
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