@InProceedings{hagsand03:low,
  author = 	 {Olof Hagsand, Ian Marsh and Kjell Hanson},
  title = 	 {Sics\textsl{o}phone: A Low-delay Internet Telephony Tool},
  OPTcrossref =  {},
  OPTkey = 	 {},
  booktitle = {29th Euromicro Conference},
  pages = 	 {189--195},
  year = 	 {2003},
  OPTeditor = 	 {Ralf Steinmetz and Andreas Mauthe},
  OPTvolume = 	 {},
  OPTnumber = 	 {},
  OPTseries = 	 {},
  OPTaddress = 	 {Belek, Turkey},
  month = 	 {Sep},
  organization = {IEEE},
  publisher = {IEEE},
  OPTnote = 	 {},
  OPTannote = 	 {}
  abstract = "The end to end delay is a critical factor in the
perceived quality of service for Voice over IP applications.
Sics\textsl{o}phone 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.",
Entryby = Im
}