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Technology Stocks : Crys*Tel Telecommunications.com, Inc. (CYSS) -- Ignore unavailable to you. Want to Upgrade?

To: Jorj X Mckie who wrote (193)	4/22/1999 12:36:00 PM
From: Chief	Read Replies (1) \| Respond to of 338

JXM, I have just visited a very interesting site which I believe will help clarify some of your concerns with CYSS and its Internet Telephony business. Please direct your web browser to : -> www.tmcnet.net and then do a search of Crys-Tel and read the findings. There are four references there. In addition, myself and a fellow investor have contacted Nokia and will produce the EXACT definitions that may clear up this issue. PLease look for that post as soon as it is scanned and prepared for this format. Incidentally, Internet Telephony Email Newsletter (www.tmcnet.com) is one of the most well read magazines in the world and I urge all urge all readers to spend some time at their site. Warm Regards, Chief

To: Jorj X Mckie who wrote (193)	4/22/1999 1:41:00 PM
From: Earl Falwell	Read Replies (1) \| Respond to of 338

JXM, If I may quote you: "The underlying message is that the Internet does not have the Quality of Service (QoS) mechanisms to provide a voice transport friendly environment. The idea of using the Internet for business telephony applications or any applications where you actually want to communicate with the other end effectively and also bill for it, is a pipedream." ********************************* In response to you comment her is some facts from Nokia. So the question remains: How does Nokia System ensure high quality voice? Their are four factors that affect voice quality. 1) bandwith (throughput) 2) delay (latency) 3) delay variation (jitter) 4) errors (packet loss) Another factor affecting voice quality in IP Telephony is the codec the IP Telephony vendor uses in their gateways. The codec compresses and decompresses. Typically, the more compression the less bandwidth it consumes but the more latency it adds. However, some are better at delivering voice quality than others due to intelligent algorithms. Based on our experiences with our customers the largest problem networks have (particularly large networks) is latency. Most networks (unless they are dial-up modem links) have kbps to spare the voice. Many of Nokia's customers offer quite acceptable voice calls over the public Internet. Jitter does not seem to be much of a problem, as we do not seem to receive many out of order packets. We have done tests over the Internet internationally that show out or order packets count for less than 1.5% of all packets, so the avantage of a larger jitter buffer is reduced substantially. However, latency (in our experience) is something that needs to be carefully controlled. Once you encounter about 350-400ms round trip, voice conversations start to become oneous, they have a "half duplex" feel to them. To that end, Nokia has prioritized latency over other optimizations. We have a small jitter buffer, as buffering packets add latency. We have optimized the flow of packets through oue system to minimize latency (to be approximatly 50 ms per gateway). Independent tests have measured the the latency between two Nokia IP Telephony Gateways connected across a TI network to be 105ms. In terms of the amount of bandwidth required for a voice call, Nokia has taken a flexible approach. There is a configurable parameter in our product that allows a network operator to optimize our system for bandwidth (use a small amount of b/w) or latency (minimize latency). Our 7.3Kbps codec takes up (dependent on the network-TI, LAN, Frame, etc. - and the configuration) between 15Kbps and 30Kbps on a network. A call can survive with less, but then the voice quality begins to be affected. In terms of packet loss, this can be mitigated. We have an intelligent function in our clients and gateways that signals the codec when we detect a lost packet(s) in a stream and the codec interpolates the lost packet(s). We have daone tests to show that 10-20% packet loss still results in very intelligible voice. How does Nokia's solution determine how a call is routed? Nokia implements a feature called "Automatic Route Selection", whereby intelligent analysis is done on a dialed number on a digit by digit basis. When enough digits have been seen to complete the decision criterion, then the call is ultimately routed and placed. This feature enables client calls to be routed via the lowest cost route to the destination PSTN network. More later, Earl