Abstract
With the advent of various web-based applications and the fourth generation (4G) access technology, there has been an exponential growth in the demand of multimedia service delivery along with speech signals in a voice over internet protocol (VoIP) setup. Need is felt to fine-tune the conventional speech codecs deployed to cater to the modern environment. This fine-tuning can be achieved by further compressing the speech signal and to utilize the available bandwidth to deliver other services. This paper presents a scalable -hybrid model of speech codec using ITU-T G.729 and db10 wavelet. The codec addresses the problem of compression of speech signal in VoIP setup. The performance comparison of the codec with the standard codec has been performed by statistical analysis of subjective, objective and quantifiable parameters of quality desirable from the codec deployed in VoIP platforms.
Highlights
In the recent years, there has been substantial standardization in the various codec deployment and access technologies for voice over internet protocol (VoIP) applications
There has been substantial standardization in the various codec deployment and access technologies for VoIP applications. These standardizations has laid the foundation of the modern 4G technology, which is a packet switched network based on internet protocol and has data rates up to 1 Gbps [1]. This compliments the general characteristics of VoIP setup of ease of launch of additional services along with the conventional telephony –voice services[2].The integration of different web-based applications in modern VoIP service had led to the various challenges of Quality of service(QoS) faced by the VoIP application
Standardization Sector (ITU-T) standardized codec known as ITU-T G.729 codec based on the principle of Conjugate Structure Algebraic Code Excited Linear Prediction (CSACELP)[5].The codec is a hybrid type codec which uses the algebraic sum of the fixed codebook gain vector and the adaptive gain to arrive at the actual codebook vector for performing the linear prediction[6].The codec operates at 8kbps and produces the highest quality of speech reproduction known as “Toll Quality” for most practical and real-time conditions and widely deployed in all voice-based operations[6]
Summary
There has been substantial standardization in the various codec deployment and access technologies for VoIP applications. Standardization Sector (ITU-T) standardized codec known as ITU-T G.729 codec based on the principle of Conjugate Structure Algebraic Code Excited Linear Prediction (CSACELP)[5].The codec is a hybrid type codec which uses the algebraic sum of the fixed codebook gain vector and the adaptive gain to arrive at the actual codebook vector for performing the linear prediction[6].The codec operates at 8kbps and produces the highest quality of speech reproduction known as “Toll Quality” for most practical and real-time conditions and widely deployed in all voice-based operations[6] This technology provides optimal complexity and is best suited for multimedia digital simultaneous voice and data communication, makes it best suitable for VoIP applications[7].The performance of this codec have been extensively tested on various environment, which resulted in concluding that the codec performs consistently with voice signals and narrowband control signals [8], international and regional speech coding standards, channel noise and over degraded transmission channels [9].
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