- 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?
