Sip components – 2N BRI Lite/Enterprise v1.1 User Manual

Page 16

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For 2N BRI gateway, quadruple the above mentioned rates (two fully duplex calls) and
add the TCP and IP header transmission rate to the result to get the resultant
transmission rate.

It is important to keep both a stable appropriate transmission rate during connection
and a small and identical transmission time per data packet in order to maintain a
high-quality voice transmission.

G.711 – this codec is used in digital telephone networks. The PCM (Pulse Code
Modulation) is used for voice signal encoding. The sampled signal is encoded
in 12 bits and then compressed using a non-linear scheme into the resultant 8
bits. Europe uses the A-law compression system while North America and
Japan obey the

-law. The resultant data flow is 64 kbps.

G.729 – this codec uses the CS-ACELP (Conjugate-Structure Algebraic-Code-
Excited Linear-Prediction) algorithm with the resultant transmission rate of 8
kbps. The speech signal is split into blocks of 10 ms each. The parameters of
these blocks are then inserted in frames of the size of 10 bytes. 2-byte
frames are generated for noise transmission.

During call set-up, a codec is selected automatically for voice transmission. 2N BRI
gateway supports the codecs included in the table above.

The type of codec to be used

depends on your VoIP network (individual devices) and your 2N BRI gateway
configuration. 2N BRI gateway is designed primarily for VoIP corporate networks and
tries to meet the opponent‟s codec requirements. If a codec is requested that is
incompatible with 2N

®

VoiceBlue Next, the call will be rejected.

The SIP and ITU-T H.323 recommended protocols are mostly used for connection
establishing, maintaining and cancelling. 2N BRI gateway uses the SIP (Session
Initiation Protocol) signalling.

Tip

In the case of separated direct connection of your SIP Proxy and 2N

®

VoiceBlue Next, use the G.711 codec to achieve a high voice quality.

SIP Components

The following components are involved in the SIP message exchange:

UAC (User Agent Client) – the terminal device client, which initiates SIP
signalling.

UAS (User Agent Server) – the terminal device server, which responds to SIP
signalling from the UAC.

UA (User Agent) – a SIP network terminal (SIP phones, gateways to other
networks, etc.), which contains the UAC and UAS.

Proxy server – receives connection requests from the UA and transfers them
to the next Proxy server if the given station is not under it administration.

Redirect server – receives connection requests, but, instead of sending them
to the called line, sends them back to the requesting device asking for where
to route the request.

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