摘要 :
In this paper, we present an approach of integrating SIP (Session Initiation Protocol) in converged multimodal/multimedia communication services. An extensible VoIPTeleserver for VoIP in SIP environment is described. It is based o...
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In this paper, we present an approach of integrating SIP (Session Initiation Protocol) in converged multimodal/multimedia communication services. An extensible VoIPTeleserver for VoIP in SIP environment is described. It is based on the concept of dialogue system and Web convergence that separates the channel dependent media resources from the application dependent service creation and hosting environment. It supports XML based service applications for multiple channels including voice, DTMF, IM and chat over IP. The loosely coupled open architecture in our approach is highly extensible. We describe the concept and structure of VoIPTeleServer used in our approach in detail, which interfaces to the VoIP world through SIP signaling and works as a broker between the VoIP SIP environment and MTIP to deliver converged communication services. A prototype of VoIPTeleServer Was implemented, and services and applications based on SIP and MTIP convergence are constructed. Special attention is given to the adverse effect of delay, jitter and packet loss for voice portal services over IP. In particular, case studies of DTMF service in voice portal under adverse channel conditions are performed. The compounding effects of multiple channel impairments to DTMF in voice portal services over IP are characterized. The potential high error rate of the DTMF service indicates that the data redundancy method as proposed in RFC 2198 is needed for DTMF in order to achieve reliable voice portal services over IP.
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摘要 :
The International Telecommunication Union (ITU) recommendations
for dual-tone multifrequency (DTMF) signaling are not met by
conventional DTMF detectors. We present an efficient DTMF detection
algorithm based on the nonuniform dis...
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The International Telecommunication Union (ITU) recommendations
for dual-tone multifrequency (DTMF) signaling are not met by
conventional DTMF detectors. We present an efficient DTMF detection
algorithm based on the nonuniform discrete Fourier transform that meets
all of the ITU recommendations. The key innovations are the use of two
sliding windows and development of sophisticated timing tests. Our
algorithm requires no buffering of input samples. To perform DTMF
detection on n telephone channels, our algorithm requires approximately
n MIPS on a digital signal processor (DSP), 75+30n words of data memory,
and 1000 words of program memory. Using the new algorithm, a single
fixed-point DSP can perform ITU-compliant DTMF detection on the 24
telephone channels of a T1 time-division multiple multiplexed
telecommunications line
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