Delay, echo, codec type, and packet loss all come into play in achieving the quality of PSTN calls in packet-based telephony.

No abstract

The mouth-to-ear delay bounds that can be tolerated for undistorted voice are well known and standardized in the ITU-T Recommendations. In this paper, similar delay bounds are determined for voice that is transported in compressed form over an IP network. More precisely, the dependency of these bounds on the low bit rate codecs used, the amount of echo control performed and the way the IP network is accessed, is investigated in detail.

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This paper discusses resource allocation and management in Differentiated Services (DiffServ) networks, particularly in the context of IP telephony. We assume that each node uses Weighted Fair Queuing (WFQ) schedulers in order to provide Quality of Service (QoS) to aggregates of traffic. All voice traffic destined for a certain output interface is aggregated into a single queue. When a voice flow traverses a node, its packets are placed in this queue; which is drained at a certain rate, determined by a weight associated with that queue. This paper shows how to set this weight such that the edge-router-to-edge-router queuing delay in the DiffServ network is statistically bounded. The nodes are modeled as M/G/1 queuing systems and a heuristic formula is used to compute a quantile of the queuing delay. This formula is compared with results derived from simulations. It is also shown how to use this result to allocate the appropriate resources in a DiffServ network and how to update and process RSVP messages at the edge of a DiffServ network in an IntServ over DiffServ scenario.