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1.	Hagsand, Olof, et al. (author) Sicsophone : A low-delay Internet telephony tool 2003 In: PROCEEDINGS OF THE 29TH EUROMICRO CONFERENCE - NEW WAVES IN SYSTEM ARCHITECTURE. - Belek, Turkey : IEEE. - 0769519962 ; , s. 189-195 Conference paper (peer-reviewed)abstract The end to end delay is a critical factor in the perceived quality of service for Voice over IP applications. Sicsophone is a complete VoIP system that couples the low level features of audio hardware with a standard jitter buffer playout algorithm. Using, the sound card directly eliminates intermediate buffering as well as providing fine control over timers needed by a soft real-time application such as VoIP A statistical based approach for inserting packets into audio buffers is used in conjunction with a scheme for inhibiting unnecessary fluctuations in the system. We also present mouth-to-ear delay measurements for selected VoIP applications and show that several hundreds of milliseconds can be saved by using the techniques described in this paper A prototype for both UNIX and Windows platforms has been implemented, demonstrating that our system adapts to network: conditions whilst maintaining low delays.

Hagsand, Olof, et al. (author)
Sicsophone : A low-delay Internet telephony tool
2003
In: PROCEEDINGS OF THE 29TH EUROMICRO CONFERENCE - NEW WAVES IN SYSTEM ARCHITECTURE. - Belek, Turkey : IEEE. - 0769519962 ; , s. 189-195
Conference paper (peer-reviewed)abstract
- The end to end delay is a critical factor in the perceived quality of service for Voice over IP applications. Sicsophone is a complete VoIP system that couples the low level features of audio hardware with a standard jitter buffer playout algorithm. Using, the sound card directly eliminates intermediate buffering as well as providing fine control over timers needed by a soft real-time application such as VoIP A statistical based approach for inserting packets into audio buffers is used in conjunction with a scheme for inhibiting unnecessary fluctuations in the system. We also present mouth-to-ear delay measurements for selected VoIP applications and show that several hundreds of milliseconds can be saved by using the techniques described in this paper A prototype for both UNIX and Windows platforms has been implemented, demonstrating that our system adapts to network: conditions whilst maintaining low delays.

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