Adaptive Rate Control for Aggregated VoIP Traffic

Sabrina, Fariza; Valin, Jean‐Marc

doi:10.1109/glocom.2008.ecp.275

Cited by 8 publications

(10 citation statements)

References 4 publications

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“…However, this algorithm does not consider the effects of end-to-end jitter buffer delay accurately and can lead to wrong decisions in case of increased delays in network that do not affect perceptual quality of the speech. [16] presents a control mechanism by dynamically adjusting the encoding bit rate based on the feedback information about the network congestion from RTCP packets and speech properties. In [17] a new quality of service control scheme was proposed by combining rate adaptation (through the use of Adaptive Multi-Rate codec (AMR) and priority marking of voice packets.…”

Section: Related Workmentioning

confidence: 99%

Quality of experience (QoE) driven adaptation scheme for voice/video over IP

et al. 2010

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Network quality of service (NQoS) of IP networks is unpredictable and impacts the quality of networked multimedia services. Adaptive voice and video schemes are therefore vital for the provision of voice over IP (VoIP) services for optimised quality of experience (QoE). Traditional adaptation schemes based on NQoS do not take perceived quality into consideration even though the user is the best judge of quality. Additionally, uncertainties inherent in NQoS parameter measurements make the design of adaptation schemes difficult and their performance suboptimal. This paper presents a QoE-driven adaptation scheme for voice and video over IP to solve the optimisation problem to provide optimal QoE for networked voice and video applications. The adaptive VoIP architecture was implemented and tested both in NS2 and in an Open IMS Core network to allow extensive simulation and test-bed evaluation. Results show that the scheme was optimally responsive to available network bandwidth and congestion for both voice and video and optimised delivered QoE for different network conditions, and is friendly to TCP traffic.

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Section: Related Workmentioning

confidence: 99%

Quality of experience (QoE) driven adaptation scheme for voice/video over IP

et al. 2010

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“…There are a number of modes Speex can operate in which relate to the data rate used to encode the voice signal, including constant bit rate (CBR), variable bit rate (VBR) and available bit rate (ABR). Speex has the capability of changing the data rate setting on a per voice frame basis [7] and the rate being used is included in the compressed voice payload. Table 1 lists details of the Speex codec and compares it with a more commonly used G.729 voice codec, also based on CELP.…”

Section: Speex: Quality and Data Ratesmentioning

confidence: 99%

A cross layer scheme for maximising the combination of wireless VoIP capacity and quality

Chen

Boreli

Iyer

2009

Proceedings of the 2009 International Conference on Wireless Communications and Mobile Computing: Connecting the World Wireless

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We present a novel mechanism for adjusting the combination of the number of simultaneous VoIP calls and voice quality, based on the monitored WiMax link parameters. We have implemented this mechanism which combines call admission control and setting of variable voice codec rate (and corresponding quality) dynamically in the Asterix PBX using the Speex voice codec. The mechanism includes dynamic parameter adjustments of currently active VoIP calls, with the capability to change quality on a per voice frame basis. It additionally limits acceptance of new calls until there is sufficient capacity to maintain a pre-defined minimum quality for the existing calls. We have also implemented the wireless link capacity estimator, which, based on the parameters measured in the receiver, estimates the data rate currently available on the wireless link. Our implementation is for a pre-WiMax service offered by UNWIRED in Australia. We show that the proposed cross layer optimisation enables a larger number of calls with improved voice quality, compared to a fixed scheme. We are currently working with a certified WiMax equipment supplier and we plan to extend the scheme to full WiMax implementation.

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“…Allocating bit rate to traffic flows based on the priority is a challenging research issue. In our preliminary work [6] we focused on cases where there is no priority. In this paper, we propose a new priority based VoIP system that employs an adaptive variable rate speech codec called Speex and a feedback based flow controller called Explicit Control Protocol (XCP) [7].…”

Section: Introductionmentioning

confidence: 99%

Priority Based Dynamic Rate Control for VoIP Traffic

Sabrina

Valin²

2009

GLOBECOM 2009 - 2009 IEEE Global Telecommunications Conference

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Abstract-This paper presents a novel mechanism for dynamic rate control of prioritised Voice Over IP (VoIP) traffic in real time. The system uses our proposed variable bit rate speech codec called Speex, which can dynamically adjust the encoding bit rate (and hence the voice quality) based on the feedback information about the network congestion, flow priority, and the instantaneous speech properties. Our extensive NS2 simulation results along with results from ITU-T standard of speech quality evaluation tool (PESQ) show that the proposed system indeed provides highest quality speech while maximising the bandwidth utilisation and reducing the network congestion. I. INTRODUCTIONVoice over IP (VoIP) has become one of the most popular IP based real-time communication applications in recent years. To support VoIP applications over the Internet, two conflicting requirements need to be met [1]. On one hand, applications are sensitive to delay, packet loss and bandwidth, so it is required to minimise the effect of network impairments on voice quality. On the other hand, since the Internet is a shared environment, resource utilisation needs to be controlled so that resource usage is optimized and congestion is avoided. This leads existing research to focus on achieving highest quality voice while both maximising the network resource utilisation and avoiding congestion.Recently, several researchers have focused on rate/quality control for multimedia flows on the Internet. Bolot et. al [2] have demonstrated a packet loss feedback based rate control scheme. However, since packet loss does not necessarily mean network congestion, algorithm based on that may not always be accurate. Mahlo et. al. [3] proposed a different approach to adapting bit-rate for VoIP flows based on an enhanced TCPfriendly rate control (TFRC) protocol that adapts the coding and packetisation to optimize the VoIP quality.In [4], researchers have also proposed a bit-rate control mechanism for VoIP application based on individual network parameters like packet loss and delay or on the predicted, perceived speech quality. In the proposed system the feedback information was sent via RTCP reports. Currently, the most common transport method for VoIP is the real-time transport protocol (RTP). Unfortunately, the cost in terms of overhead is large. Considering that most speech codecs use a frame size of 20 ms, the sum of the IP, UDP and RTP headers (20+8+12=40 bytes), sent 50 times per second represents 16 kbit/s. This means that typically at least half of the traffic in a VoIP conversation is headers. For many applications, we

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Adaptive Rate Control for Aggregated VoIP Traffic

Cited by 8 publications

References 4 publications

Quality of experience (QoE) driven adaptation scheme for voice/video over IP

Quality of experience (QoE) driven adaptation scheme for voice/video over IP

A cross layer scheme for maximising the combination of wireless VoIP capacity and quality

Priority Based Dynamic Rate Control for VoIP Traffic

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