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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

1 2 3 4 5 6 7 ... 36 37

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