Performance Analysis of Hybrid Codecs G. 711 and G. 729 over Signaling Protocols H. 323 and SIP (original) (raw)
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Journal of electrical electronics engineering, 2024
Advancements in internet technology have enabled the integration of different traffic types i.e. data, video, and voice into a single network. This technology offers many benefits but also presents some challenges. Real-time traffic services such as VoIP require a certain Quality of Service (QoS) which cannot be guaranteed on the Internet therefore, key performance metrics become all the more important. The choice of codecs and queuing techniques becomes crucial for ensuring optimal performance, especially in networks with diverse traffic types. This research therefore compared the effects of the combinations of these tools (i.e. queuing techniques and codec schemes) on the quality of VoIP. A simulation approach using the OPNET Modeler 14.5 tool has been used to simulate a network supporting three different types of traffic namely: FTP traffic, Video conferencing traffic, and VoIP traffic. While maintaining the same topology and traffic of the network, different types of codec schemes and queuing techniques have been tested through the measurement of parameters such as delay and throughput. The custom Queuing technique showed the best performance overall while FIFO suffered the highest delay. The graphs were observed to follow the same pattern regardless of the codec scheme used however, G729 performed the better of the two as it received higher amounts of voice traffic and slightly lower delays compared to G711.
Performance evaluation of voice over IP using multiple audio codec schemes
ARPN journal of engineering and applied sciences, 2015
The evolution of Voice over IP (VoIP) has made it one of the most popular applications over the wired/wireless Internet system due to its flexibility in technology integration and low cost of services. Telco and service operators have used the communication resources to optimize the VoIP architecture in order to provide better quality of service (QoS) to end consumers. The VoIP is a delay-sensitive traffic which requires minimum delay for general applications and minimum loss ratio for specific applications as the key QoS performance parameters. This paper compares the end-to-end (e2e) QoS performance parameters of VoIP codec schemes against multiple traffic connections transmitted over the Internet system. Background traffics are included in the simulations to closely match the real-world Internet scenario. Simulations analysis of bidirectional VoIP communications are done from the network layer perspective to compare the QoS performances of G.711, G.729A, G.723.1 and GSM.AMR codec...
Performance Analysis between H.323 and SIP over VoIP
2019
There are a number of protocols that may be employed in order to provide the Voice over IP VoIP communication services. In VoIP system, H.323 and Session Initiation Protocol SIP are the two major standards. Both of these signaling protocols provide mechanisms for multimedia teleconferencing services. Although the two protocols architecture is quite similar, they have many differences. This system presents Voice Video over IP communication and summarizes the differences and performance of two major VoIP protocols, H.323 and SIP according to the packet delay variation, jitter, packet loss, and Packet end to end delay. It is found that both of them are non interoperable, approaching each other, their focus and applicability is still different. In this paper, the system is designed and configured by Graphical Network Simulator GNS3 and analyzed performance by Opnet Modeler Simulation. Thet Zaw Aye "Performance Analysis between H.323 and SIP over VoIP" Published in Internationa...
Audio Codecs in VoIPv6: A Performance Analysis
International Journal of Computer Network and Information Security, 2014
Audio communications in IP based networks have been revolutionized by the introduction of VoIP applications. High cost-efficiency has made VoIP to be the communication means in today's world; and this trend is anticipated to be continued on an ongoing basis. The performance of VoIP significantly depends on the efficiency of the audio codecs used in any communication scenario which make the study on the performance issues of audio codecs in VoIP applications worth investigating. IPv6 is the new version of IP, which will gradually replace the current IPv4 as the transition from IPv4 to IPv6 is already in place. This demands the scrutiny of the audio codecs being used in IPv4 to be tested for their compatibility in IPv6 in terms of desired performance. This paper presents the study on the performance of selected audio codecs that are widely used in VoIPv4. G.711, G.729A and G.723.1 codecs were chosen for the study in VoIPv6 based scenarios presented in this paper. The selected audio codecs were applied in IPv6 based voice communication network scenarios to determine their performance efficiency by observing various QoS parameters. The study was done by means of simulation using OPNET.
Performance Evaluation of the QoS for VoIP using Different CODECS
Voice over Internet Protocol (VoIP) service is growing very fast and supported by many applications. Its interactive nature makes it very attractive service. VoIP requires a precise level of quality to be utilized. Quality of Service (QoS) is determined by factors like jitter, traffic sent, traffic received and end-to-end delay. In this paper, we study the performance of different scheduling schemes, like: FIFO, PQ, and WFQ for different codec formats. The implementation of the schemes was carried out using OPNET. VoIP service is deployed using the internet implementing the Resource Reservation Protocol (RSVP). The paper discusses the results through a number of figures for the jitter, end-to-end delay and the traffic sent and received. Figures show the different scheduling schemes PQ, WFQ and FIFO with different codec formats, G.711, G.729A and G.723.15 codec formats. I. INTRODUCTION Nowadays, very huge amounts of voice traffic are transferred between millions of people across the world using different social media applications. Using VoIP over the Internet connection, we should be aware about the quality of the VoIP service. VoIP service requires a precise level of quality to be utilized. The end user perception of the quality is determined by subjective testing as a function of the network impairments such as delay, jitter, packet loss, and blocking probability. The amount of impairment introduced by a packet network depends on the particular QoS mechanism implemented [1] Quality of Service (QoS) is determined by factors like the delay the packet delay variation (jitter), and the data loss rate [2]. The greatest technical problem in supporting multimedia services over IP is that real-time traffic must reach its destination within a preset time interval (delay) and with some tolerance of the delay variation (jitter). This is difficult because the original UDP/IP operates on a best-effort basis and permits dropping of packets on the way to a destination [3]. The simulation model was done using OPNET Modeler [4] [5]. OPNET has gained considerable popularity in academia as it is being offered free of charge to academic institutions. That has given OPNET an edge over DES NS2 in both market place and academia [6]. In this paper, we studied the performance of the most popular scheduling schemes, like: First-In First-Out (FIFO), priority Queuing (PQ), and Weighted Fair Queuing (WFQ). A comparison is carried out between different codecs (G.711, G.729A and G.723.15) which are the most appropriate to improve QoS for VoIP. The rest of the paper is organized as follows. Section II presents a typical WAN network topology that uses RSVP protocol to be used as a case study for deploying VoIP service. Section III describes the VoIP service and its parameters. Section IV presents the OPNET-based simulation approach for deploying VoIP service. Section V describes the results and analysis of the simulation study. Then section VI conclusion.
Analysis Performance VoIP Codecs over WiMAX Access Network
WiMAX Technology is also one of the emerging wireless technology that provide us high speed mobile data and telecommunication services. WiMAX is the first wireless standard that allows the service provider or user to choose a codec based on the application Voice over Internet Protocol (VoIP) is an internet technology for the transmission of voice and multimedia over the Internet Protocol (IP) based network, especially the Internet. It has been widely in use as a communication protocol to replace traditional telephone technology, Public Switched Telephone Network (PSTN). As OPNET 14.5 provides simulated real-life environment, we have chosen as the platform OPNET simulation to study the performance of all of this work. The parameters used to evaluate the network throughput, jitter and packet and to and delay. Performance of VoIP codec is used to analyze traffic on the network wimax.
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