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October 2013 – Ed.1.0

EQUITEL E400 Family – Control and communications protocol

Page 23

Chapter 3. Audio communication using RTP and G.711-A

The audio communication within EQUITEL IP intercom system (E401 AND E451) uses the RTP protocol with the

G.711-A encoding.

This communication implies two processes: encoding and decoding.

With regards to encoding, the equipments E401 and E451 send the RTP stream to the indicated IP address and

port in the communication establishment phase. Depending on the unit configuration, the communication can

be established in different ways:

• Proprietary mode: using EQUITEL proprietary protocol, described in Chapter 2.

• SIP mode with PBX: using the SIP protocol and a PBX, detailed in Chapter 4.

• P2P-SIP mode: using the SIP protocol. There is no need of a PBX, as explained in Chapter 4.

As to decoding, the systems of the family E400 always answer to the input UDP port 30000.

In this chapter we shall describe the encoding and decoding processes used in our systems.

3.1 Encoding

The process of encoding and encapsulating the audio is shown in the following block diagram:

Figure 11. G711 coding and RTP encapsulation

Roughly, the process is as follows:

The input audio is digitized at 8000 samples per second. Each sample must have a minimum resolution of 12

bits

(6)

These audio samples are stored in a buffer with capacity for up to 160 samples, the equivalent to 20

milliseconds of audio.

Every time we have a 160 samples packet, the G.711-A coding is applied, consisting basically in converting

each 12bits PCM sample in an 8 bits data according to the following table:

(6) If we take a computer as an example, it is usual to take samples at 16 bits and with the proper treatment they are reduced till 12 bits.

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Equitel E451 Bedienungsanleitung Seite 26