Tab. 1 - Compression in VoIP
2.3 Transport and signaling protocol in VoIP network
For telephony over IP based network we need not only transmit audio data also we need
transmit a drive information about connection and formation and unformation of connection.
The several voice data are transmitted mostly via RTP protocol. Mostly aplied signaling
(driving) protocol are:
• H.323 by ITU-T
• SIP (Session Initiate Protocol) by IETF (RFC 2443)
• MGCP (Media Gateway to Media Controller Protocol) by Cisco and Bellcore
2.3.1 RTP (Real Time Transport Protocol)
The sampled and compressed data are encapsulated into TCP/IP stack. VoIP data packets
live in RTP (Real-Time Transport Protocol) or RSVP packets which are inside UDP-IP packets.
Tab. 2 -RM-OSI reference model
Processing of sampled signal
Formats Note
Speed [kbit/s]
Recommendation
PCM (mu-law, A-law) Uncompressed data
64 kbit/s (8 kHz *
8bit)
G.711
ADPCM
Transmit of
difference between
samples
32 G.721 G.726 G.727
LD-CELP linear prediction 16 G.728
CS-ACELP linear prediction 8 G.729
MP-MLQ
True speech
compression
6,3 G.723.1
LPC-10 linear prediction 2,5 -
GSM VSELP based 9,6 – 14,4 -
RM-OSI layer model
Num. of layer
Name of Layer Protocol
7 Application
6 Presentation
VoIP data, H.323,
SIP or MGCP
5 Session RTP
4 Transport UDP
3 Network IP
2 Link Frame (Eth. ATM..)
1 Physical Medium
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