ZyXEL Communications P-2602HW(L) Series - Manual (Page 156)

P-2602H(W)(L)-DxA Series User’s Guide
156 Chapter 11 Voice
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.
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 53 SIP > SIP Settings
LABEL DESCRIPTION