* [[http://www.csie.ntu.edu.tw/~acpang/course/voip_2005_fall/|網頁]]
  * 12/05 停課 && 12/10 13:00 補課

  * Why payload type in RTP is necessary (even with SIP)?
    * due to RED?
  * How to calculate jitter?
  * for each received RTP packet, you know when it was sent (RTP Timestamp) and when it is received.. so.. just do it. also check RFC 1889 A.8
  * Why source description packet needed even with SIP
  * sip registers with multiplue server
    * failure tolerance
    * multiple registration: multiple locations (follow-me) ringing

===== final =====
  * elaborate on the principles of integration on SIP & QoS mechanism
  * STUN
    * describe how STUN works for most NATs
    * describe why STUN can not work on Symmetric NAT
    * describe Binding Lifetime Determination in STUN
  * describe the adv. of separation of media and call control
  * (SIP-to-PSTN gateway based on MGCP) Draw network arch. and describe message flow for SIP terminal calling a telephone on PSTN.
  * describe issues discussed in IETF SIGTRAN
  * describe concepts of Softswitch, compare it with existing Hard/Circulted-switch
  * <i>in what scenario will Softswitch be a selection for VoIP service?</i>
  * Design a VoIP Netowrk
    * blocking probability? calculate? 
    * factors does the required bandwidth for a single call depend on?