ZyXEL Communications LTE3311 series - Technical Reference

Chapter 17 Voice over IP
LTE3311 Series User’s Guide
127
The following table describes the labels in this screen.
17.8 Technical Reference
This section contains background material relevant to the VoIP screens.
VoIP
VoIP is the sending of voice signals over Internet Protocol. This allows you to make phone calls and
send faxes over the Internet at a fraction of the cost of using the traditional circuit-switched
telephone network. You can also use servers to run telephone service applications like PBX services
and voice mail. Internet Telephony Service Provider (ITSP) companies provide VoIP service.
Circuit-switched telephone networks require 64 kilobits per second (Kbps) in each direction to
handle a telephone call. VoIP can use advanced voice coding techniques with compression to reduce
the required bandwidth.
SIP
The Session Initiation Protocol (SIP) is an application-layer control (signaling) protocol that handles
the setting up, altering and tearing down of voice and multimedia sessions over the Internet.
Table 59 Application > Voice over IP > SIP Conf.: Codec Conf.
LABEL DESCRIPTION
Codec Priority Select the type of voice coder/decoder (codec) that you want the LTE3311 to use.
G.711 provides high voice quality but requires more bandwidth (64 kbps). G.711 is the
default codec used by phone companies and digital handsets.
• G.711a-law is typically used in Europe.
• G.711u-law is typically used in North America and Japan.
G.726-32 operates at 32 kbps.
G.729 provides good sound quality and reduces the required bandwidth to 8 kbps.
The LTE3311 must use the same codec as the peer. When two SIP devices start a SIP
session, they must agree on a codec.
Priority1
Select the LTE3311’s first choice for voice coder/decoder.
Priority2 Select the LTE3311’s second choice for voice coder/decoder.
Priority3 Select the LTE3311’s third choice for voice coder/decoder.
Priority4 Select the LTE3311’s fourth choice for voice coder/decoder.
SIP Packet Length
for G.711&G.729
Select to change the voice payload size for VoIP packets. If you increase the voice
payload size, the bandwidth will be reduced.
Voice VAD Select Enable if the LTE3311 should stop transmitting when you are not speaking. This
reduces the bandwidth the LTE3311 uses.
Apply
Click this to save your changes and to apply them to the LTE3311.
Cancel Click this to set every field in this screen to its last-saved value.
Back Click this to return to the previous screen.