A Survey of Bandwidth Optimization Techniques and Patterns in VoIP Services and Applications
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paper_title: Performance measurement for voice services in heterogeneous wired networks

paper_content:

Voice over IP (VoIP) service providers implementing different protocols in their core networks are required to coordinate in order to provision, support and deliver voice services to end clients. This paper presents a performance measurement experience for the transport of voice services across multiple heterogeneous wired networks. Our test results show that Multiple Protocol Label Switching (MPLS) protocol slightly outperforms IP in delivering voice services. Moreover, the type of call signaling protocol used, tunneling category adopted and the type of Virtual Private Network (VPN) implemented affect the overall quality and performance of voice services in terms of jitter and delay.
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paper_title: Improving QoS of VoWLAN via cross-layer interaction approach

paper_content:

Voice over Internet Protocol (VoIP) is a transmission technology that allows voice to be delivered over the Internet with support of Internet Protocol. Recently, the integration of VoIP and Wireless Local Area Network (WLAN) has become popular, driven by mobility requirements of the users. However, WLAN architecture was primarily designed to carry data and not for voice. Hence, there is a need to enhance the quality of voice transmission over the WLAN. Several metrics such as delay, jitter, and packet lost have impact on the quality of service (QoS) of the VoIP system. Minimizing total transmission delay, jitter and packet lost would mean improving the QoS of VoIP. In this paper, a cross-layer interaction approach is proposed to improve the quality of voice delivery, hence the QoS of VoIP. The approach in achieving this is by introducing a communication agent between Data Link layer and Transport layer, by which congestion misinterpretation by the Transport layer could now be eliminated. In particular, the IP protocol of Transport layer will be utilized to communicate with the agent to improve the QoS of the VoIP system.
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paper_title: Voice -over -IP (Voip) Bandwidth Optimization: A Survey of Schemes and Techniques

paper_content:

Voice over Internet Protocol (VoIP) is a voice communication system that proffers the transmission and reception of audio (voice) signal and (or) data over the internet. One very crucial problem in a VoIP network is inefficient bandwidth utilization which in most case is caused by header overhead resulting from the attachment of a big (40 bytes) to small payloads (10 to 30 bytes). Given the need to reduce the amount of bandwidth being wasted or perhaps improving the utilization of bandwidth which ultimately could enhance network performance and VoIP quality of service, several schemes and techniques have been proposed. The improvement and (or) optimization of bandwidth can be achieved through; packet header reduction and defines a process of multiplexing and de-multiplexing packet headers to curb overflows. Silence suppression offer a technique of repressing the silent portions (packets) in a voice conversation using Voice Activity Detection algorithm, as a result, the transmission rate during the inactive periods of speech is reduced, and thus, the average transmission rate can be reduced. Voice/ Packet Header compression is considered the most productive of all the technique, and it presents an event whereby VoIP packets are compressed from the conventional 40 bytes of size to a smaller byte size (2 bytes in the best case). Bandwidth save is reached using compression and decompression codecs of varying data and bit rates. It is envisaged that an improvement in the performance of codecs would yield a better result in terms of optimizing bandwidth in a VoIP network.
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paper_title: Internet telephony: going like crazy

paper_content:

Interet telephony is possibly the fastest-growing part of communications today. This article discusses what exactly it is, who needs it, and how it works. Internet telephony, or voice over Internet protocol (VoIP), is the provision of phone service over the Internet. But in sharp contrast with conventional telephony, it carries voice traffic as data packets over a packet-switched data network instead of as a synchronous stream of binary data over a circuit-switched, time-division multiplexed (TDM) voice network. There are some substantial benefits (as well as some sticky problems) to the scheme, which is why companies and individuals are finding it increasingly attractive.
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paper_title: Optimizing Voip Using A Cross Layer Call Admission Control Scheme

paper_content:

Deploying wireless campus network becomes popular in many world universities for the services that are provided. However, it suffers from different issues such as low VoIP network capacity, network congestion effect on VoIP QoS and WLAN multi rate issue due to linkadaptation technique. In this paper a cross layer call admission control (CCAC) scheme is proposed to reduce the effects of these problems on VoWLAN based on monitoring RTCPRR(Real Time Control Protocol ReceiverReport) that provides the QoS level for VoIP and monitoring the MAC layer for any change in the data rate. If the QoS level degrades due to one of the aforementioned reasons, a considerable change in the packet size or the codec type will be the solution. A wireless campus network issimulated using OPNET 14.5 modeler and many scenarios are modeled to improve this proposed scheme.
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paper_title: Performance Optimizations for Deploying VoIP Services in Mesh Networks

paper_content:

In the recent past, there has been a tremendous increase in the popularity of VoIP services as a result of huge growth in broadband access. The same voice-over-Internet protocol (VoIP) service poses new challenges when deployed over a wireless mesh network, while enabling users to make voice calls using WiFi phones. Packet losses and delay due to interference in a multiple-hop mesh network with limited capacity can significantly degrade the end-to-end VoIP call quality. In this work, we discuss the basic requirements for efficient deployment of VoIP services over a mesh network. We present and evaluate practical optimizing techniques that can enhance the network capacity, maintain the VoIP quality and handle user mobility efficiently. Extensive experiments conducted on a real testbed and ns-2 provide insights into the performance issues and demonstrate the level of improvement that can be obtained by the proposed techniques. Specifically, we find that packet aggregation along with header compression can increase the number of supported VoIP calls in a multihop network by 2-3 times. The proposed fast path switching is highly effective in maintaining the VoIP quality. Our fast handoff scheme achieves almost negligible disruption during calls to roaming clients
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paper_title: RTP payload multiplexing between IP telephony gateways

paper_content:

We describe a new method to multiplex a number of low bit rate audio streams into a single RTP stream between IP telephony gateways. Audio frames from different users are assembled into a multiplexed RTP payload thus reducing the overhead of RTP/UDP/IP headers. To maintain distinction between the multiplexed audio streams sharing a single RTP stream, a 2-byte mini-header is added to each frame of an audio stream. A unique channel identifier is associated between the peer gateway entities by means of signaling procedures. Major applications of this method will be in the areas of IP telephony gateways that interconnect PSTN/PBX systems and cellular access networks. Results have indicated that the overhead is reduced by 50% to 80% depending on the size of the audio frame. Other advantages of the proposed scheme include fewer UDP/IP connections between gateways, a reduced possibility of congestion at intermediate IP routers, and a reduction of processing delays in multiprotocol label switching (MPLS), differentiated service (diff-serv) and IP security (IPsec) environments.
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paper_title: Towards an efficient header compression scheme to improve VoIP over wireless mesh networks

paper_content:

Header compression techniques can improve the speech quality of VoIP applications. However, its deployment over wireless mesh environments may not offer much improvement, since the wireless and multi-hop characteristics of such a network impose a high packet loss rate. In this work, we propose a new approach for header compression, using packet aggregation to cooperate with the header compression scheme. Simulation results show that the new cooperative approach can decrease significantly the packet loss, impacting positively on the speech quality.
---
paper_title: Performance Optimizations for Deploying VoIP Services in Mesh Networks

paper_content:

In the recent past, there has been a tremendous increase in the popularity of VoIP services as a result of huge growth in broadband access. The same voice-over-Internet protocol (VoIP) service poses new challenges when deployed over a wireless mesh network, while enabling users to make voice calls using WiFi phones. Packet losses and delay due to interference in a multiple-hop mesh network with limited capacity can significantly degrade the end-to-end VoIP call quality. In this work, we discuss the basic requirements for efficient deployment of VoIP services over a mesh network. We present and evaluate practical optimizing techniques that can enhance the network capacity, maintain the VoIP quality and handle user mobility efficiently. Extensive experiments conducted on a real testbed and ns-2 provide insights into the performance issues and demonstrate the level of improvement that can be obtained by the proposed techniques. Specifically, we find that packet aggregation along with header compression can increase the number of supported VoIP calls in a multihop network by 2-3 times. The proposed fast path switching is highly effective in maintaining the VoIP quality. Our fast handoff scheme achieves almost negligible disruption during calls to roaming clients
---
paper_title: Towards an efficient header compression scheme to improve VoIP over wireless mesh networks

paper_content:

Header compression techniques can improve the speech quality of VoIP applications. However, its deployment over wireless mesh environments may not offer much improvement, since the wireless and multi-hop characteristics of such a network impose a high packet loss rate. In this work, we propose a new approach for header compression, using packet aggregation to cooperate with the header compression scheme. Simulation results show that the new cooperative approach can decrease significantly the packet loss, impacting positively on the speech quality.
---
paper_title: Performance Optimizations for Deploying VoIP Services in Mesh Networks

paper_content:

In the recent past, there has been a tremendous increase in the popularity of VoIP services as a result of huge growth in broadband access. The same voice-over-Internet protocol (VoIP) service poses new challenges when deployed over a wireless mesh network, while enabling users to make voice calls using WiFi phones. Packet losses and delay due to interference in a multiple-hop mesh network with limited capacity can significantly degrade the end-to-end VoIP call quality. In this work, we discuss the basic requirements for efficient deployment of VoIP services over a mesh network. We present and evaluate practical optimizing techniques that can enhance the network capacity, maintain the VoIP quality and handle user mobility efficiently. Extensive experiments conducted on a real testbed and ns-2 provide insights into the performance issues and demonstrate the level of improvement that can be obtained by the proposed techniques. Specifically, we find that packet aggregation along with header compression can increase the number of supported VoIP calls in a multihop network by 2-3 times. The proposed fast path switching is highly effective in maintaining the VoIP quality. Our fast handoff scheme achieves almost negligible disruption during calls to roaming clients
---
paper_title: Voice -over -IP (Voip) Bandwidth Optimization: A Survey of Schemes and Techniques

paper_content:

Voice over Internet Protocol (VoIP) is a voice communication system that proffers the transmission and reception of audio (voice) signal and (or) data over the internet. One very crucial problem in a VoIP network is inefficient bandwidth utilization which in most case is caused by header overhead resulting from the attachment of a big (40 bytes) to small payloads (10 to 30 bytes). Given the need to reduce the amount of bandwidth being wasted or perhaps improving the utilization of bandwidth which ultimately could enhance network performance and VoIP quality of service, several schemes and techniques have been proposed. The improvement and (or) optimization of bandwidth can be achieved through; packet header reduction and defines a process of multiplexing and de-multiplexing packet headers to curb overflows. Silence suppression offer a technique of repressing the silent portions (packets) in a voice conversation using Voice Activity Detection algorithm, as a result, the transmission rate during the inactive periods of speech is reduced, and thus, the average transmission rate can be reduced. Voice/ Packet Header compression is considered the most productive of all the technique, and it presents an event whereby VoIP packets are compressed from the conventional 40 bytes of size to a smaller byte size (2 bytes in the best case). Bandwidth save is reached using compression and decompression codecs of varying data and bit rates. It is envisaged that an improvement in the performance of codecs would yield a better result in terms of optimizing bandwidth in a VoIP network.
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paper_title: Efficient transport of voice over IP over cellular links

paper_content:

Internet and cellular technologies are merging to decrease transport costs and enable IP-based applications to hundreds of millions of cellular users. Real time voice services will continue to be dominant in future cellular systems and to reduce system costs it is desirable to base all services, including voice, on IP. However, the radio spectrum must be efficiently used to provide mass-market services at reasonable prices, but Internet protocol headers are large in size. Moreover, data discarding policies are poorly suited to the error-proneness of cellular links. Protocol overhead for IP-based voice services can be reduced by compressing headers, e.g. with the compressed real time protocol (CRTP). CRTP, however, does not perform well over cellular links since it is not robust in the face of errors. ROCCO, a header compression scheme suitable for cellular systems, delivers more packets to the user than CRTP at realistic error rates with a better average compression ratio. Voice services in cellular systems should also be able to use damaged speech data. This can be done using the UDP Lite protocol which provides a more flexible checksumming policy that enables delivery of data possibly usable to the application. This paper shows that CRTP/UDP fails to deliver sufficient service quality at the error rates of a cost-efficient cellular system and that ROCCO combined with UDP Lite can provide an IP-based voice service over such cellular systems.
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paper_title: Towards an efficient header compression scheme to improve VoIP over wireless mesh networks

paper_content:

Header compression techniques can improve the speech quality of VoIP applications. However, its deployment over wireless mesh environments may not offer much improvement, since the wireless and multi-hop characteristics of such a network impose a high packet loss rate. In this work, we propose a new approach for header compression, using packet aggregation to cooperate with the header compression scheme. Simulation results show that the new cooperative approach can decrease significantly the packet loss, impacting positively on the speech quality.
---
paper_title: Proposal of a method of voice stream multiplexing for IP telephony systems

paper_content:

IP telephony systems are expected to be deployed worldwide in the near future because of their potential for integrating the multimedia communication infrastructure over IP networks. Phone-to-phone connection over an IP network via IP telephony gateways (IP-GWs) is a key feature of the system. In an IP telephony system, a low-bit-rate voice codec is used to improve bandwidth efficiency. However, due to the packet transfer method over the IP network, it is necessary to add packet headers, including IP, UDP, and RTP headers, which increases the header overhead and thus decreases transfer efficiency. Moreover, because there will be large numbers of short voice packets flowing into the IP network, the load on the Internet will increase. We propose voice stream multiplexing between IP-GWs to solve these problems. In this scheme, multiple voice streams are connected between a pair of IPCWs, enabling multiplexed voice stream transfer. The voice stream multiplexing mechanism can reduce the header overhead as well as decrease the number of voice packets. The voice stream multiplexing we propose is to concatenate RTP packets destined for the same IP-GW at a multiplexing interval period into a single UDP packet. The advantage of this method is that no new additional header is required and the current well-defined H.323 and RTP standards can be applied with minimum changes. We implemented and tested the system.
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paper_title: A Multiplexing Scheme for H.323 Voice-Over-IP Applications

paper_content:

Voice communications such as telephony are delay sensitive. Existing voice-over-IP (VoIP) applications transmit voice data in packets of very small size to minimize packetization delay, causing very inefficient use of network bandwidth. This paper proposes a multiplexing scheme for improving the bandwidth efficiency of existing VoIP applications. By installing a multiplexer in an H.323 proxy, voice packets from multiple sources are combined into one IP packet for transmission. A demultiplexer at the receiver-end proxy restores the original voice packets before delivering them to the end-user applications. Results show that the multiplexing scheme can increase bandwidth efficiency by as much as 300%. The multiplexing scheme is fully compatible with existing H.323-compliant VoIP applications and can be readily deployed.
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paper_title: RTP payload multiplexing between IP telephony gateways

paper_content:

We describe a new method to multiplex a number of low bit rate audio streams into a single RTP stream between IP telephony gateways. Audio frames from different users are assembled into a multiplexed RTP payload thus reducing the overhead of RTP/UDP/IP headers. To maintain distinction between the multiplexed audio streams sharing a single RTP stream, a 2-byte mini-header is added to each frame of an audio stream. A unique channel identifier is associated between the peer gateway entities by means of signaling procedures. Major applications of this method will be in the areas of IP telephony gateways that interconnect PSTN/PBX systems and cellular access networks. Results have indicated that the overhead is reduced by 50% to 80% depending on the size of the audio frame. Other advantages of the proposed scheme include fewer UDP/IP connections between gateways, a reduced possibility of congestion at intermediate IP routers, and a reduction of processing delays in multiprotocol label switching (MPLS), differentiated service (diff-serv) and IP security (IPsec) environments.
---
paper_title: Modified silence suppression algorithms and their performance tests

paper_content:

In packet voice communication systems, silence suppression algorithms are employed to achieve bandwidth efficiency by suppressing the inactive part of the speech. In this paper, we consider a Nortel fixed-point silence suppression algorithm to investigate means of reducing the computational complexity with a little or no loss in the performance. In view of the compact structure and superior performance of the existing Nortel algorithm, certain modifications are introduced without altering the structure of the original design. Based on these modifications, two algorithms are proposed. Results from objective and subjective quality tests show that the performance of each of the two modified algorithms is comparable to that of the Nortel algorithm. As for the average computational complexity, simulation tests based on TMS320C5402 DSK show that reduction of about 40% and 22% can be achieved for the two modified algorithms.
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paper_title: Comparison of voice activity detection algorithms for VoIP

paper_content:

We discuss techniques for voice activity detection (VAD) for voice over Internet Protocol (VoIP). VAD aids in saving the bandwidth requirement of a voice session, thereby increasing the bandwidth efficiently. We compare the quality of speech, level of compression and computational complexity for three time-domain and three frequency-domain VAD algorithms. Implementation of time-domain algorithms is computationally simple. However, better speech quality is obtained with the frequency-domain algorithms. A comparison of the merits and demerits along with the subjective quality of speech after removal of silence periods is presented for all the algorithms. A quantitative measurement of speech quality for different algorithms is also presented.
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paper_title: VAD techniques for real-time speech transmission on the Internet

paper_content:

We discuss techniques for voice activity detection (VAD) for voice over Internet Protocol (VoIP). VAD aids in reducing the bandwidth requirement of a voice session, thereby using bandwidth efficiently. Such a scheme would be implemented in the application layer. Thus the VAD is independent of the lower layers in the network stack (see Flood, J.E., "Telecommunications Switching - Traffic and Networks", Prentice Hall India). We compare four time-domain VAD algorithms in terms of speech quality, compression level and computational complexity. A comparison of the relative merits and demerits along with the subjective quality of speech after the pruning of silence periods is presented for all the algorithms. A quantitative measurement of speech quality for different algorithms is also presented.
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paper_title: Modified silence suppression algorithms and their performance tests

paper_content:

In packet voice communication systems, silence suppression algorithms are employed to achieve bandwidth efficiency by suppressing the inactive part of the speech. In this paper, we consider a Nortel fixed-point silence suppression algorithm to investigate means of reducing the computational complexity with a little or no loss in the performance. In view of the compact structure and superior performance of the existing Nortel algorithm, certain modifications are introduced without altering the structure of the original design. Based on these modifications, two algorithms are proposed. Results from objective and subjective quality tests show that the performance of each of the two modified algorithms is comparable to that of the Nortel algorithm. As for the average computational complexity, simulation tests based on TMS320C5402 DSK show that reduction of about 40% and 22% can be achieved for the two modified algorithms.
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Title: A Survey of Bandwidth Optimization Techniques and Patterns in VoIP Services and Applications

Section 1: INTRODUCTION
Description 1: This section provides an introduction to the paper, outlining the importance and growth of VoIP technology, its benefits, and the challenges related to bandwidth inefficiency.

Section 2: VOIP BANDWIDTH OPTIMIZATION: SPURS AND TECHNIQUES
Description 2: This section discusses the need for bandwidth optimization in VoIP networks and details various techniques used for optimizing bandwidth.

Section 2.1: Why Optimization?
Description 2.1: This subsection explains the reasons why bandwidth optimization is crucial for VoIP systems, including competitive industry demands and operational inefficiencies.

Section 2.2: Optimization Techniques
Description 2.2: This subsection introduces the major techniques for improving bandwidth utilization in VoIP networks/services, such as header compression/suppression, packet header reduction, and silence suppression.

Section 2.2.1: Voice / Packet Header Compression
Description 2.2.1: This subsection explores how voice packets can be compressed to reduce the default header size using specific codecs and algorithms.

Section 2.2.1.1: Compression with Aggregation
Description 2.2.1.1: This subsection discusses the combination of header compression with packet aggregation to eliminate redundant information and improve bandwidth efficiency.

Section 2.2.1.2: Compressed Real-time Transport Protocol (cRTP)
Description 2.2.1.2: This subsection details the use of Compressed Real-time Transport Protocol to compress headers of voice packets for better bandwidth utilization.

Section 2.2.1.3: Robust Checksum-based header Compression (ROCCO) Compression
Description 2.2.1.3: This subsection explains how ROCCO addresses the weaknesses of cRTP by reducing packet loss during decompression and improving header compression performance.

Section 2.2.1.4: CRTP / ROCCO UDP Lite Integration
Description 2.2.1.4: This subsection discusses the integration of UDP Lite protocol with cRTP and ROCCO to improve packet transmission efficiency.

Section 2.2.1.5: Robust Header Compression (RoHC)
Description 2.2.1.5: This subsection covers how the RoHC protocol efficiently saves bandwidth by integrating properties of earlier compression schemes.

Section 2.2.2: Packet Header Reduction
Description 2.2.2: This subsection discusses processes like multiplexing and de-multiplexing to reduce packet header sizes and overall overhead.

Section 2.2.2.1: Multiplexing
Description 2.2.2.1: This subsection describes how combining multiple payloads in one header can significantly cut overhead and save bandwidth.

Section 2.2.2.1.a: RTP Multiplexing Techniques and Applications
Description 2.2.2.1.a: This subsection details various RTP multiplexing techniques and their applications in reducing bandwidth usage.

Section 2.2.2.2: Delta-Multiplexing (De-Multiplexing)
Description 2.2.2.2: This subsection outlines the delta-multiplexing technique for combining header overhead reductions and payload size deductions for better bandwidth utilization.

Section 2.2.2.2.a: De(Multiplexing) Applications
Description 2.2.2.2.a: This subsection provides examples of how delta-multiplexing is applied in different network settings to improve bandwidth efficiency.

Section 2.2.3: Silence Suppression
Description 2.2.3: This subsection describes how silence suppression techniques can save bandwidth by eliminating transmission during silent periods in voice communication.

Section 3: CONCLUSION AND FUTURE WORK
Description 3: This section provides a summary of the findings on bandwidth optimization techniques, evaluates their effectiveness, and suggests areas for future research and improvement.