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G.711
G.726
G.729
G723.1
Algorithm
PCM
ADPCM
CS-ACELP
ACELP
Transmission Rate
64
32
8
5.3
MOS
4.1
3.85
3.92
3.56
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constant bandwidth, guaranteed time of packet delivery (also called jitter) and
correct sequence are necessary for successful voice transmission. We need not
worry about delivery of any packet during voice transmission because mathematical
methods used for voice signal coding and decoding can make approximation when a
packet has not been delivered. Thus, we can use the UDP for voice stream
transmission, which has no acknowledgement of delivered packets, but in any case
we need a protocol that is responsible for voice coding, jitter, sequence order and
bandwidth. This protocol is called RTP (Realtime Transport Protocol) and is widely
used for voice transmission in modern VoIP networks.
UAC (User agent client) client in the terminal that initiates SIP signalling
UAS (User agent server) server in the terminal that responds to the SIP
signalling from the UAC
Proxy server receives connection requests from the UA and transfers them
to another proxy server if the particular station is not in its administration
Redirect server receives connection requests and sends them back to the
requester including destination data instead of sending them to the calling
party
Location Server receives registration requests from the UA and updates the
terminal database with them.
All server sections (Proxy, Redirect, Location) are typically available on a single
physical machine called proxy server, which is responsible for client database
maintenance, connection establishing, maintenance and termination, and call
directing.
Basic messages sent in the SIP environment
Answers to SIP messages are in the digital format like in the http protocol. Here are
the most important ones:
5XX server error (500 Server Internal Error, 501 not implemented)