course:internettelephony
Why payload type in RTP is necessary (even with SIP)?
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
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
course/internettelephony.txt · Last modified: 2007/03/06 13:54 (external edit)