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Guaranteed Delay for Voice Traffic in an Integrated Packet-Switched Network

IP.com Disclosure Number: IPCOM000038543D
Original Publication Date: 1987-Feb-01
Included in the Prior Art Database: 2005-Jan-31
Document File: 3 page(s) / 19K

Publishing Venue

IBM

Related People

Gopal, PM: AUTHOR [+2]

Abstract

Packet-switched transmission of voice traffic can result in variable network delays for different packets belonging to the same speaker. This can lead to discontinuities in the output speech if the packets are delivered to the destination as they arrive. Consequently, additional buffering of voice packets is done at the destination in order to smooth out the variation in delay. Specifically, a target playout time is chosen for each packet, and packets not arriving within this time are discarded. The target playout time can be chosen based on the network delay distribution such that the constraints on the delay and packet loss are met. This requires that the destination keep track of network conditions and adapt to the changes from time to time.

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Guaranteed Delay for Voice Traffic in an Integrated Packet-Switched Network

Packet-switched transmission of voice traffic can result in variable network delays for different packets belonging to the same speaker. This can lead to discontinuities in the output speech if the packets are delivered to the destination as they arrive. Consequently, additional buffering of voice packets is done at the destination in order to smooth out the variation in delay.

Specifically, a target playout time is chosen for each packet, and packets not arriving within this time are discarded. The target playout time can be chosen based on the network delay distribution such that the constraints on the delay and packet loss are met. This requires that the destination keep track of network conditions and adapt to the changes from time to time. An alternative strategy is to guarantee that the delay does not exceed a given value. This makes the playout process much simpler in terms of choosing the target playout time. Moreover, the destination does not have to worry about changes in the network conditions once a call is established. We present a procedure by means of which bounded delay can be achieved. This procedure can be used with or without speech activity detection (SAD), as long as the total offered traffic does not exceed the link capacities. For simplicity, we assume that the network carries only voice traffic. We further assume that all speakers use the same coding rate and packet sizes. The network delay without SAD is fairly small [*] under the assumption of random arrival of calls. However, the worst-case delay can be very large. A packet belonging to a particular speaker may have to wait for transmissions from all other speakers before it reaches the destination. This can happen if at every hop, packet arrivals from all other speakers occurred just before the packet (under consideration) arrives. In order to reduce the worst-case delay, one can limit the number of speakers sharing each link based on the maximum number of hops travelled by any call. However, this approach is unduly restrictive for the calls travelling less than the maximum number of hops. An approach which is promising is based on non-preemptive priority scheduling. Specifically, each voice call going over a link has a unique priority level associated with it. This assignment is made at call setup time. This ensures that the packets belonging to any call can only be delayed by packets from higher priority calls (plus at most one lower priority packet because of the non-preemptive nature of priority). However, due to the delay variations introduced by call initiations and terminations at earlier hops, it is possible that at some hop, more than one packet from a higher priority level ma...