ZyXEL P-2024 User Guide - Page 75

LABEL, DESCRIPTION, Start Port, End Port, G.711A, G.729, Secondary Compression Type, RFC 2833

Page 75 highlights

Chapter 6 SIP Each field is described in the following table. Table 22 VoIP > SIP > SIP Settings > Advanced LABEL DESCRIPTION SIP Account This field displays the number of the SIP account you see in this screen. SIP Server Settings URL Type Select whether or not to include the SIP service domain name when the P-2024 sends the SIP number. SIP - include the SIP service domain name TEL - do not include the SIP service domain name Expiration Duration Enter the number of seconds your SIP account is registered with the SIP register server before it is deleted. The P-2024 automatically tries to re-register your SIP account when one-half of this time has passed. (The SIP register server might have a different expiration.) Register Re- Enter the number of seconds the P-2024 waits before it tries again to register the send timer SIP account, if the first try failed or if there is no response. Session Expires Enter the number of seconds the conversation can last before the call is automatically disconnected. Usually, when one-half of this time has passed, the P2024 or the other party updates this timer to prevent this from happening. Min-SE Enter the minimum number of seconds the P-2024 accepts for a session expiration time when it receives a request to start a SIP session. If the request has a shorter time, the P-2024 rejects it. RTP Port Range Start Port End Port Enter the listening port number(s) for RTP traffic, if your VoIP service provider gave you this information. Otherwise, keep the default values. To enter one port number, enter the port number in both the Start Port and the End Port fields. To enter a range of ports, • enter the port number at the beginning of the range in the Start Port field • enter the port number at the end of the range in the End Port field. Voice Compression Select the type of voice coder/decoder (codec) that you want the P-2024 to use. G.711 provides higher voice quality but requires more bandwidth (64 kbps). • G.711A is typically used in Europe. • G.711u is typically used in North America and Japan. In contrast, G.729 only requires 8 kbps. The P-2024 must use the same codec as the peer. When two SIP devices start a SIP session, they must agree on a codec. Primary Select the P-2024's first choice for voice coder/decoder. Compression Type Secondary Select the P-2024's second choice for voice coder/decoder. Select None if you Compression only want the P-2024 to accept the first choice. Type Third Compression Type This field is disabled if Secondary Compression Type is None. Select the P-2024's third choice for voice coder/decoder. Select None if you only want the P-2024 to accept the first or second choice. DTMF Mode Control how the P-2024 handles the tones that your telephone makes when you push its buttons. You should use the same mode your VoIP service provider uses. RFC 2833 - send the DTMF tones in RTP packets PCM - send the DTMF tones in the voice data stream. This method works best when you are using a codec that does not use compression (like G.711). Codecs that use compression (like G.729) can distort the tones. SIP INFO - send the DTMF tones in SIP messages P-2024 User's Guide 75

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Chapter 6 SIP
P-2024 User’s Guide
75
Each field is described in the following table.
Table 22
VoIP > SIP > SIP Settings > Advanced
LABEL
DESCRIPTION
SIP Account
This field displays the number of the SIP account you see in this screen.
SIP Server
Settings
URL Type
Select whether or not to include the SIP service domain name when the P-2024
sends the SIP number.
SIP
- include the SIP service domain name
TEL
- do not include the SIP service domain name
Expiration
Duration
Enter the number of seconds your SIP account is registered with the SIP register
server before it is deleted. The P-2024 automatically tries to re-register your SIP
account when one-half of this time has passed. (The SIP register server might
have a different expiration.)
Register Re-
send timer
Enter the number of seconds the P-2024 waits before it tries again to register the
SIP account, if the first try failed or if there is no response.
Session
Expires
Enter the number of seconds the conversation can last before the call is
automatically disconnected. Usually, when one-half of this time has passed, the P-
2024 or the other party updates this timer to prevent this from happening.
Min-SE
Enter the minimum number of seconds the P-2024 accepts for a session
expiration time when it receives a request to start a SIP session. If the request has
a shorter time, the P-2024 rejects it.
RTP Port Range
Start Port
End Port
Enter the listening port number(s) for RTP traffic, if your VoIP service provider
gave you this information. Otherwise, keep the default values.
To enter one port number, enter the port number in both the
Start Port
and the
End Port
fields.
To enter a range of ports,
enter the port number at the beginning of the range in the
Start Port
field
enter the port number at the end of the range in the
End Port
field.
Voice
Compression
Select the type of voice coder/decoder (codec) that you want the P-2024 to use.
G.711 provides higher voice quality but requires more bandwidth (64 kbps).
G.711A
is typically used in Europe.
G.711u
is typically used in North America and Japan.
In contrast,
G.729
only requires 8 kbps.
The P-2024 must use the same codec as the peer. When two SIP devices start a
SIP session, they must agree on a codec.
Primary
Compression
Type
Select the P-2024’s first choice for voice coder/decoder.
Secondary
Compression
Type
Select the P-2024’s second choice for voice coder/decoder. Select
None
if you
only want the P-2024 to accept the first choice.
Third
Compression
Type
This field is disabled if
Secondary Compression Type
is
None
.
Select the P-2024’s third choice for voice coder/decoder. Select
None
if you only
want the P-2024 to accept the first or second choice.
DTMF Mode
Control how the P-2024 handles the tones that your telephone makes when you
push its buttons. You should use the same mode your VoIP service provider uses.
RFC 2833
- send the DTMF tones in RTP packets
PCM
- send the DTMF tones in the voice data stream. This method works best
when you are using a codec that does not use compression (like G.711). Codecs
that use compression (like G.729) can distort the tones.
SIP INFO
- send the DTMF tones in SIP messages