Arcom Control Systems

News Release from: Spirit DSP
Edited by the Electronicstalk Editorial Team on 9 December 2004

VoIP codec aims to improve speech at any rate

Spirit has developed a new-generation VoIP codec for IP networks.

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Spirit has developed a new-generation VoIP codec for IP networks. High speech quality is a must in modern VoIP systems, but standard ITU codecs like G.729 and G.723 are becoming obsolete. Unlike alternative IP codecs on the market, Spirit IP-Multi Rate delivers high-quality voice at any rate in the range from 4 to 80Kbit/s.

The codec works in both narrowband and wideband mode and is highly adaptive to network conditions thanks to automatic rate selection.

Moreover the codec is transcoding-free in multiparty conferences saving server resources and ensuring high quality.

Thanks to scalable coding, all sent packets are ranged according to the contained information importance and transmitted to the client side independently.

High-priority packets are transmitted first of all, so that information of interest is always delivered without artefacts.

Lower priority packets are also delivered if the connection quality is good enough.

IP-MR is designed to overcome the adverse network conditions of packet loss up to 30% and more and variable network errors and delays.

IP-MR ensures that the bandwidth consumed is no more than the amount of useful information in the speech, thus ensuring optimal traffic and channel usage during the whole conversation.

'Spirit IP-MR is optimal for both peer-to-peer and multipoint conferencing, being adapted for most efficient tandem-free IP conferencing', said Andrew Sviridenko, founding CEO of Spirit.

'The user is able to speak in full duplex mode and to hear high quality speech even in terms of low connection quality of his remote colleague'.

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