ZyXEL Communications P-2602R-DxA Series User Manual

P-2602R/RL-DxA Series User’s Guide

Chapter 10 Voice

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When you make a VoIP call using SIP, the RTP (Real Time Transport Protocol) is used to 
handle voice data transfer. See RFC 1889 for details on RTP.

Pulse Code Modulation (PCM) measures analog signal amplitudes at regular time intervals 
and converts them into bits.

A codec (coder/decoder) codes analog voice signals into digital signals and decodes the digital 
signals back into analog voice signals. The ZyXEL Device supports the following codecs.

G.711 is a Pulse Code Modulation (PCM) waveform codec. G.711 provides very good sound 
quality but requires 64kbps of bandwidth.

G.729 is an Analysis-by-Synthesis (AbS) hybrid waveform codec that uses a filter based on 
information about how the human vocal tract produces sounds. G.729 provides good sound 
quality and reduces the required bandwidth to 8kbps.

SIP Service 

Domain

Enter the SIP service domain name. In the full SIP URI, this is the part after the @ 

symbol.  You can use up to 127 printable ASCII Extended set characters.

Send Caller ID

Select this if you want to send identification when you make VoIP phone calls. 

Clear this if you do not want to send identification.

Authentication

User Name

Enter the user name for registering this SIP account, exactly as it was given to you. 

You can use up to 95 printable ASCII characters.

Password

Enter the user name for registering this SIP account, exactly as it was given to you. 

You can use up to 95 printable ASCII Extended set characters.

Apply

Click this to save your changes and to apply them to the ZyXEL Device.

Cancel

Click this to set every field in this screen to its last-saved value.

Advanced Setup

Click this to edit the advanced settings for this SIP account. The Advanced SIP 

Setup screen appears.

Table 37   SIP > SIP Settings