scholarly journals Using the Packet Header Redundant Fields to Improve VoIP Bandwidth Utilization

2021 ◽  
Author(s):  
Manjur Kolhar
Keyword(s):  
Internet Protocol ◽  
Ip Networks ◽  
Voice Over Internet Protocol ◽  
Call Rate ◽  
Bandwidth Utilization ◽  
Network Bandwidth ◽  
Packet Header ◽  
Two Factors ◽  
Contrast Technique

Timeworn telecommunication are progressively being substituted by a new one that run over IP networks, which is recognized as voice over internet protocol (VoIP). VoIP has a number of qualities (e.g., inexpensive call rate), which make it progressively widespread in the telecommunication domain. However, VoIP faces plentiful obstacles that slow its growth. One of the major obstacles is poorly utilizing the network bandwidth. A number of techniques have been offered to handle this obstacle, including packet multiplexing techniques. This paper designs an original multiplexing techniques, called packet multiplexing and carrier header (PM-CH), to decrease the quantity of the bandwidth consumed by VoIP. PM-CH protect the bandwidth by multiplexing the packets in a header and using the Timestamp field in the RTP header. The achievement of the PM-CH technique was examined depends on connection capacity and payload shortening. Simulation outcomes show that the PM-CH technique outperforms the contrast technique in the two factors. For instance, the PM-CH technique’s connection capacity outperforms the comparable technique by 58.9% when the connection bandwidth is 1000 kbps. Consequently, the PM-CH technique attains its objective of reducing the unexploited bandwidth caused by VoIP.

scholarly journals Network Steganography System Using Covert Channel For LSBS Stego Data On VOIP Communication

2019 ◽  
Vol 8 (5C) ◽  
pp. 1448-1449
Keyword(s):  
Internet Protocol ◽  
Ip Networks ◽  
Covert Channel ◽  
Voice Over Internet Protocol ◽  
Two Stage ◽  
Art And Science ◽  
Large Capacity ◽  
Network Steganography ◽  
Two Stages ◽  
Least Significant Bits

Voice over Internet Protocol (VoIP) is a realtime service that enables voice conversations using IP networks, Steganography is the art and science of hiding secret messages in other messages that makes the secret messages are unknowable. VoIP allows as a medium cover carrier on steganography to provide the security of confidential messages. This study built by two-stage of steganography system using Least Significant Bits and Covert Channel that use image and VoIP communication as a medium in the hiding of messages. End of this study, the system will go through the process of performance and functionality testing. The results showed that the method of Covert Channel to the Payload field has the hiding of large capacity that are 56bit per packet, and by using two stages method of steganography can make a Steganogram extraction or analysis more difficult to do..

scholarly journals Zeroize: A New Method to Improve the Utilization of 5G Networks When Running VoIP over IPv6

2021 ◽  
Vol 4 (4) ◽  
pp. 72
Author(s):  
Manjur Kolhar
Keyword(s):  
Internet Protocol ◽  
New Method ◽  
Voice Over Internet Protocol ◽  
5G Networks ◽  
Telecommunication Sector ◽  
The Core ◽  
Packet Header ◽  
Ipv6 Protocol ◽  
A New Technique ◽  
Voice Data

5G technology is spreading extremely quickly. Many services, including Voice Over Internet Protocol (VoIP), have utilized the features of 5G technology to improve their performance. VoIP service is gradually ruling the telecommunication sector due to its various advantages (e.g., free calls). However, VoIP service wastes a substantial share of the VoIP 5G network’s bandwidth due to its lengthy packet header. For instance, the share of the packet header from bandwidth and channel time reaches 85.7% of VoIP 5G networks when using the IPv6 protocol. VoIP designers are exerting considerable efforts to solve this issue. This paper contributes to these efforts by designing a new technique named Zeroize (zero sizes). The core of the Zeroize technique is based on utilizing the unnecessary fields of the IPv6 protocol header to keep the packet payload (voice data), thereby reducing or “zeroizing” the payload of the VoIP packet. The Zeroize technique substantially reduces the expanded bandwidth of VoIP 5G networks, which is reflected in the wasted channel time. The results show that the Zeroize technique reduces the wasted bandwidth by 20% with the G.723.1 codec. Therefore, this technique successfully reduces the bandwidth and channel time of VoIP 5G networks when using the IPv6 protocol.

Comparison of QoS Performance Over WLAN, VoIP4 and VoIP6

2015 ◽  
Vol 2 (11) ◽  
pp. 42
Author(s):  
Esra Musbah Mohammed Musbah ◽  
Khalid Hamed Bilal ◽  
Amin Babiker A. Nabi Mustafa
Keyword(s):  
Performance Analysis ◽  
Internet Protocol ◽  
Voice Over Internet Protocol ◽  
Ip Network ◽  
Qos Parameters ◽  
Software Program ◽  
Jitter Delay ◽  
Qos Performance ◽  
Delay Variation ◽  
A Performance

VoIP stands for voice over internet protocol. It is one of the most widely used technologies. It enables users to send and transmit media over IP network. The transition from IPv4 to IPv6 provides many benefits for internet IPv6 is more efficient than IPv4. This paper presents a performance analysis of VoIP over WLAN using IPv4 and IPv6 and OPNET software program to simulate the protocols and to investigate the QoS parameters such as jitter, delay variation, packet send, and packet received and throughputs for IP4 and IP6 and compare between them.

PENGARUH PENGGUNAAN CODEC STANDART ITU G.729 TERHADAP SISTEM KOMUNIKASI VOIP

2021 ◽  
Vol 17 (1) ◽  
pp. 11-22
Author(s):  
Wahyu Adi Prijono
Keyword(s):  
Communication System ◽  
Communication Systems ◽  
Internet Protocol ◽  
Voice Over Internet Protocol ◽  
Data Communications ◽  
Voice Communication ◽  
Internet Telephony ◽  
Voice Signal ◽  
Ip Network ◽  
Packet Switch

Voice over Internet Protocol (VoIP) is a technology that is capable of passing voice traffic, in the form of packets through the network Internet Protocol (IP). IP network itself is a data communications network based packet-switch. The voice signal before experiencing bundled voice coding or format conversion of sound into digital form that can be passed over an IP network. Telephony, Internet telephony, or termed VoIP (Voice Over Internet Protocol.This communication system use VoIP (Voice over Internet Protocol), ie voice calls over data services (internet). This communication was developed using Android-based devices. Based on characteristics, android devices are open source, so users do not need to have a license to be able to have android-based devices. In addition, the android device that must be connected to a SIP (Session Iniation Protocol) is a data service that can be done with a paid subscription of the user of the operator using a conventional pulse. Telecommunications designed will use a hybrid system, the merger between VoIP communications with data communications GSM network. With basic calculations where Coding standards G 729, is a standard that can be used for voice communication system through data networks with rate of 8 Kbps. The implementation of the G729 codec is effect on communication systems VOIP.

scholarly journals Rancang Bangun Voice Over Internet Protocol dan GSM Gateway Berbasis Raspberry Pi

2019 ◽  
Vol 5 (1) ◽  
pp. 55-64
Author(s):  
Ardi Windiarto ◽  
Kholilatul Wardani
Keyword(s):  
Packet Loss ◽  
Internet Protocol ◽  
Quality Of Services ◽  
Line Of Sight ◽  
Raspberry Pi ◽  
Voice Over Internet Protocol ◽  
Non Line Of Sight

Makalah ini membahas desain layanan jaringan komunikasi VoIP Server menggunakan Raspberry Pi sebagai alat komunikasi wireless. VoIP server berbasis Raspberry Pi menggunakan sistem operasi RasPBX. Di dalam sistem operasi RasPBX sudah ada software asterisk yang berfungsi sebagai softswicth. Client VoIP menggunakan zoiper sebagai softphone. Alat ini dilengkapi dengan fitur GSM gateway yaitu fitur yang dapat menghubungkan jaringan VoIP ke jaringan GSM. Fitur GSM gateway ini menggunakan modem GSM sebagai jembatan yang menghubungkan jaringan VoIP dengan jaringan GSM. Persentase keberhasilan panggilan VoIP ke VoIP, VoIP ke GSM, dan GSM ke VoIP mencapai 100%. Berdasarkan hasil pengujian Quality of services (QoS) pada panggilan VoIP ke GSM, dihasilkan rata-rata delay sebesar 12,11 ms yang termasuk dalam kategori kualitas baik, Troughput sebesar 0,151, jitter sebesar 0,052 ms yang termasuk dalam kategori kualitas baik, dan packet loss sebesar 0% yang termasuk dalam kategori kualitas sangat baik. Jangkauan maksimal antara client VoIP ke server agar komunikasi berjalan dengan baik adalah 100 meter dalam kondisi Line Of Sight (LOS). Pengujian dengan jarak 25 m dalam kondisi Non Line Of Sight (NLOS), masih menghasilkan komunikasi yang baik. Berdasarkan hasil pengujian kuisioner dari 30 pengguna, dihasilkan nilai MOS 3,88 yang termasuk dalam kategori kualitas cukup baik.

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