Rtp; Pulse Code Modulation; Voice Coding; 156 - ZyXEL Communications Network Device P-2602 User Manual

P-2602H(W)(L)-DxA Series User's Guide
Table 53 SIP > SIP Settings
LABEL
SIP Service
Domain
Send Caller ID
Authentication
User Name
Password
Apply
Cancel
Advanced Setup

11.3.1 RTP

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.

11.4 Pulse Code Modulation

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

11.5 Voice Coding

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.
11.5.1 G.711
G.711 is a Pulse Code Modulation (PCM) waveform codec. G.711 provides very good sound
quality but requires 64kbps of bandwidth.
11.5.2 G.729
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.

156

DESCRIPTION
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.
Select this if you want to send identification when you make VoIP phone calls.
Clear this if you do not want to send identification.
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.
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.
Click this to save your changes and to apply them to the ZyXEL Device.
Click this to set every field in this screen to its last-saved value.
Click this to edit the advanced settings for this SIP account. The Advanced SIP
Setup screen appears.
Chapter 11 Voice