Hagsand, Olof and Marsh, Ian and Hanson, Kjell (2002) Sicsophone: A Low-Delay Internet Telephony Tool. [SICS Report]
The end to end delay is a critical factor in the perceived quality of service for Voice over IP applications. The described solution is a complete system-level platform and complements QoS work in the network and application areas. We describe a VoIP system that couples the low level features of audio hardware with a 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 give comparisons for the performance of the playout algorithm against idealised playout conditions. 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 (NT and 9X) has been implemented, demonstrating that our system adapts to network conditions whilst maintaining low delays.
|Item Type:||SICS Report|
|Uncontrolled Keywords:||Low Delay, VoIP, buffer playout|
|Deposited By:||Vicki Carleson|
|Deposited On:||29 Oct 2007|
|Last Modified:||18 Nov 2009 16:04|
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