Performance of adaptive multirate (AMR) voice in GSM and WCDMA

Holma, Harri; Melero, Juan; Vainio, J.; Halonen, Timo; Makinen, J.

doi:10.1109/vetecs.2003.1208774

Cited by 15 publications

(8 citation statements)

References 2 publications

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“…The link and codec adaptation mechanisms [4,5] were one step ahead to construct QoS solution for VoIP over multirate wireless link. These mechanisms are providing resilience and robustness in cellular networks [6]. Hence, it is required to provide MANETs with the same capabilities.…”

Section: State Of the Artmentioning

confidence: 99%

“…In this process, the node monitors VoIP activity and codec changes from the upper layers and then, if appropriate, applies adaptation function "1" or goes to the next monitoring process. Output: HELLO interval of (1) Begin (2) if LVA = then (3) read codec parameters: sample size, sample interval (4) if sample size is high then (5) Increase (6) else if sample interval is high then 7decrease (8) end if (9) end if (10) end if (iii) Monitoring Routing Performance-Adaptation Function "Default. " This process occurs when adaptation function "1" or "2" is applied.…”

Section: (I) Monitoring Local Voip Activity-(lva-) Adaptationmentioning

confidence: 99%

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A New QoS Management Scheme for VoIP Application over Wireless Ad Hoc Networks

Brak

Benhaddou

2014

Journal of Computer Networks and Communications

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Nowadays, mobile ad hoc networks (MANETs) have to support new applications including VoIP (voice over IP) that impose stringent QoS (quality of service) constraints and requirements. However, VoIP applications make a very inefficient use of the MANET resources. Our work represents a first step toward improving aspects at the network layer by addressing issues from the standpoint of adaptation, claiming that effective adaptation of routing parameters can enhance VoIP quality. The most important contribution is the adaptive OLSR-VA algorithm which provides an integrated environment where VoIP activity is constantly detected and routing parameters are adapted in order to meet the application requirements. To investigate the performance advantage achieved by such algorithm, a number of realistic simulations (MANET scenarios) are performed under different conditions. The most important observation is that performance is satisfactory in terms of the perceived voice quality.

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Section: State Of the Artmentioning

confidence: 99%

Section: (I) Monitoring Local Voip Activity-(lva-) Adaptationmentioning

confidence: 99%

A New QoS Management Scheme for VoIP Application over Wireless Ad Hoc Networks

Brak

Benhaddou

2014

Journal of Computer Networks and Communications

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“…This adaptation can be used to increase GSM spectral efficiency while maintaining the voice quality in erroneous radio channels [4]. The GSM in-band signaling supports adaptation between four active codec modes.…”

Section: Gsm Link Adaptationmentioning

confidence: 99%

Source signal based rate adaptation for GSM AMR speech codec

Makinen

Vainio

2004

International Conference on Information Technology: Coding and Computing, 2004. Proceedings. ITCC 2004.

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Adaptive Multi-Rate (AMR) codec [1] was standardised for GSM in 1999. AMR offers substantial improvement over previous GSM speech codecs [6] in error robustness by adapting speech and channel coding depending on channel conditions. However, current standard do not exploit the multi-rate capability of AMR codec in source signal based adaptation that would optimise the average bit-rate vs. quality trade-off. This paper presents a source signal based rate adaptation algorithm for AMR codec in GSM system. Together with fast power control, it can be used to increase the system capacity and further increase the robustness of GSM AMR codec.

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“…Since its initial deployment in the early 90's, several enhancements to GSM have allowed a substantial improvement in its spectral efficiency. In particular frequency-hopping and adaptive multi-rate (AMR) [2] are well established techniques which deserve to be mentioned. GERAN is still evolving though, and there are several new features aimed at increasing system capacity, such as interference rejection combining (IRC) [3], [4], dynamic frequency and channel allocation (DFCA) [5], single-antenna interference cancellation (SAIC) [6], [7], source-adaptive AMR (SA-AMR) [8], [9] and 8-PSK [10], [11] speech channels.…”

Section: Introductionmentioning

confidence: 99%

GERAN Evolution for Increased Speech Capacity

Barreto¹,

Garcia²,

Souza³

2007

2007 IEEE 65th Vehicular Technology Conference - VTC2007-Spring

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In order to accommodate the growing traffic demand in GSM cellular networks, several features are continuously being developed to improve the voice capacity of GSM/EDGE Radio Access network (GERAN). In this paper we investigate the capacity of a GSM network considering the use of some of these features, namely, interference rejection combining (IRC), single-antenna interference cancelation (SAIC), sourceadaptive AMR (SA-AMR), dynamic frequency and channel allocation (DFCA), and 8-PSK voice channels. These techniques have been previously investigated individually in the literature, but here we focus on the system-level impact of the interactions among them. We show that the proper combination of these features increases the spectral efficiency of speech channels in GERAN.

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Performance of adaptive multirate (AMR) voice in GSM and WCDMA

Cited by 15 publications

References 2 publications

A New QoS Management Scheme for VoIP Application over Wireless Ad Hoc Networks

A New QoS Management Scheme for VoIP Application over Wireless Ad Hoc Networks

Source signal based rate adaptation for GSM AMR speech codec

GERAN Evolution for Increased Speech Capacity

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