• 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?
 
course/internettelephony.txt · Last modified: 2007/03/06 13:54 (external edit)     Back to top
Recent changes RSS feed Creative Commons License Donate Powered by PHP Valid XHTML 1.0 Valid CSS Driven by DokuWiki