VoIP: A comprehensive survey on a promising technology

Voice over IP (VoIP) is a real time application that allows transmitting voice through the Internet network. Recently there has been amazing progress in this field, mainly due to the development of voice codecs that react appropriately under conditions of packet loss, and the improvement of intelligent jitter buffers that perform better under conditions of variable inter packet delay. In addition, there are other factors that indirectly benefited VoIP. Today, computer networks are faster due to the advances in hardware and breakthrough algorithms. As a result, the quality of VoIP calls has improved considerably. However, the quality of VoIP calls under extreme conditions of packet loss still remains a major problem that needs to be addressed for the next generation of VoIP services. This thesis concentrates in making an analysis of the effects that network impairments, such as: delay, jitter, and packet loss have in the quality of VoIP calls and approaches to solve this problem. Finally, we analyze the impact of introducing forward error correction (FEC) Piggybacking and Reed Solomon codes for VoIP. To measure the mean opinion score of VoIP calls we develop an application based on the E-Model, and utilize perceptual evaluation of speech quality (PESQ).

/pdf/an-analysis-of-the-mos-under-conditions-of-delay-jitter-and-2dw7wbcn6e.pdf

An Analysis of the MOS under Conditions of Delay, Jitter and Packet Loss and an Analysis of the Impact of Introducing Piggybacking and Reed Solomon FEC for VOIP

Dynamic Multipoint IPsec VPNs (DMVPN) is IPsec VPN solution in Cisco IOS Software. This paper discusses QoS assurance in DMVPN spoke-to-spoke deployment, when using different routing protocols. Investigation of tunnel creation influence to multiple flow packets' delay in spoke-to-spoke connection by means of OPNET MODELER 10.5 was done here. The performance analysis made, allowed us to ascertain, which of routing protocols, used for DMVPN connection ensures the least delay value. It is determined that the least one was obtained when using RIP (Routing Information Protocol) and EIGRP (Interior Gateway Routing Protocol) were applied when establishing the route.

Route creation influence on DMVPN QoS

In this paper we present the results from an experimental study of the call capacity of an IEEE 802.11b network when using VoIP telephony. Experiments include increasing the number of VoIP users in a wireless test-bed and increasing the level of background traffic until network saturation occurs. The experiments allow us to perform an analysis of the access point (AP) buffer dynamics. Results show that the network is capable of supporting up to 16 VoIP stations (STAs). Due to the operation of the IEEE 802.11 medium access control (MAC) mechanism, the AP acts as a bottleneck for all traffic destined for wireless STAs, in that significant delays can be incurred by VoIP packets which can lead to a poor perceived QoS by users. Consequently the performance of the AP downlink is the critical component in determining VoIP call capacity. We also show that large peak loads occur during periods of double-talk effectively reducing the capacity of the network.

/pdf/an-experimental-analysis-of-the-call-capacity-of-ieee-802-12flae1gyb.pdf

An Experimental Analysis of the Call Capacity of IEEE 802.11b Wireless Local Area Networks for VoIP Telephony

The quality of VoIP communication relies significantly on the network that transports voice packets because this network does not usually guarantee available bandwidth, delay, and loss that are critical for real-time voice traffic. The solution proposed here is to manage a voice-over-IP stream dynamically, changing encoding parameters as needed to assure quality. The paper proposes an adaptive-rate control algorithm that establishes interaction between a VoIP sender and a receiver, and manages voice quality in real-time. Simulations demonstrate that the system provides better average communications quality than traditional fixed-rate VoIP.

Adaptive Speech Quality Management in Voice-over-IP Communications

This paper evaluates suitable voice-data length in IP packets for the adjustment of VoIP network systems. Based on measurements In a real environment, we examined the voice-quality level while varying the voice-data length of IP packets under various network conditions. We found that a VoIP system with long-voice data has high-transmission efficiency but there is a high deterioration in the voice-quality level in an inferior network. We also discovered that a VoIP system with short-voice data is tolerant to packet losses and preserves voice quality. Based on these results, we propose a VoIP system that sets the voice-data length of IP packets according to dynamically changing network QoS conditions to achieve both high-transmission efficiency and stable voice quality.

T. Takenaga

Papers

Study on appropriate voice data length of IP packets for VoIP network adjustment