INTERNET DRAFT Timer

This document is an Internet-Draft. Internet-Drafts are working documents of the Internet Engineering Task Force (IETF), its areas, and its working groups. Note that other groups may also distribute working documents as Internet-Drafts. Internet-Drafts are draft documents valid for a maximum of six months and may be updated, replaced, or obsoleted by other documents at any time. It is inappropriate to use Internet-Drafts as reference material or to cite them other than as "work in progress". To view the entire list of current Internet-Drafts, please check the "lid-abstracts.txt" listing contained in the Internet Drafts Shadow Directories on ftp.is.co.za (Africa), ftp.nordu.net (Europe), Abstract RTP, the Real Time Transport Protocol, has gained widespread acceptance as the transport protocol for voice and video on the Internet. It provides services such as timestamping, sequence numbering , and payload identiication. It also contains a control component, the Real Time Control Protocol (RTCP), which is used for loose session control, QoS reporting, and media synchronization , among other functions. The RTP speciication describes an algorithm for determining the RTCP packet transmission rate at a host participating in a multicast RTP session. This algorithm was designed to allow RTP to be used in sessions with anywhere from one to a million members. However, we have discovered several problems with this algorithm when used with very large groups with rapidly changing group membership. One problem is the ood of RTCP packets which occurs when many users join a multicast RTP session at nearly the same time. To solve this problem, we present a novel adaptive timer algorithm called reconsideration. We present a mathematical analysis of this algorithm, and demonstrate that it performs extremely well, reducing the congestion problem by several orders of magnitude. We also back up these results with simulation.