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Answer Posted / pavan_mustyala
I will give example of end-to-end call in a SIP network.
SIP is Session Initiation Protocol which is signalling
protocol mainly taking care call establishment between end
parties. Initially a series signalling messages flow
between the end parties for connection establishment and
capabilities negotiation. INVITE,RINGING,ACKNOWLEDGE,etc.
After successful negotiation and acknowledgements, RTP
session is initiated between end parties and the RTP
packets carry the voice information. The voice data is
sampled at regular intervals of time, and
compressed/decompressed according to the CODEC used. Some
of the CODECs are G711, G723,G729,etc which differ in the
compression ratio thereby effecting the effective BandWidth
(BW). G711 uses more BW where G729 uses less compared to
G711. But QOS gets effected with type of CODEC used which
is the most challenging area in VOIP currently. G711
provides better QOS compared to G723 which provides better
QOS compared to G729.
In summary, RTP in conjunction with signalling protocols
like SIP paves the way for VOIP.
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