Comp 346/488: Intro to Telecommunications

Tuesdays and Thursdays, 6:00-9:15, Lewis Towers 410

Class 4
AICN readings:
    An Overview of Networks
Optional reading (Stallings 7th -- 9th editions)
Chapters 3, 4 and 5

Homework 1 is now on Sakai. It is due Friday, July 17.


The next blog post by Whisper Systems, on Client-Side Audio Quality, goes on to say

All VoIP solutions contend with packet switched networks that were not designed to transmit real-time media streams. Packet latency and packet loss are the principal manifestations of this reality, and the jitter buffer is their canonical solution.

They then describe an adaptive-jitter-buffer algorithm, where the buffer can change size. That's a bit of a trick, as that means slowing down or speeding up the audio. In order to do this at all, one must know that sometimes voice can be stretched (eg by pausing), or shortened. For voice, the resultant distortion is generally acceptable; for music, it is likely not.



Telecom terminology: pots.html

Asterisk: configuring a phone
    Ekiga demo with headset
    PBX Terminology: asterisk_basics.html