User's Manual

Chapter 14 VoIP 102
PRACK (RFC 3262) RFC 3262 defines a mechanism to provide reliable transmission of SIP provisional
response messages, which convey information on the processing progress of the
request. This uses the option tag 100rel and the Provisional Response
ACKnowledgement (PRACK) method.
Select Suppo
rted or Req
uired to have the Router include a SIP Require/Supported
header field with the option tag 100rel in all INVITE requests. When the Router
receives a SIP response message indicating that the phone it called is ringing, the
Router sends a PRACK message to have both sides confirm the message is received.
If you select Supported, the peer device supports the option tag 100rel to send
provisional responses reliably.
If you select Req
uired, the peer device requires the option tag 100rel to send
provisional responses reliably.
Select Disabled to t
u
rn off this function.
DNS SRV Enabled
(RFC
3263)
Select this to have the Router query your ISP’s DNS server for a list of any available SIP
servers that it maintains. This is useful if your static SIP server experiences difficulties,
making it hard for your IP phone users to make SIP calls.
Session Timer
(RFC 4028)
Select this to have the Router support RFC 4028.
This makes sure that SIP sessions do not hang and the SI
P line can always be available
for use.
RTP Port Range Enter the listening port number(s) for RTP traffic provided by your VoIP service
pro
vider.
Start Port
End Port
If your VoIP service provider gave you this information. Otherwise, keep the default
values.
To enter one port number, enter the port number in the Start Port
and End Port
fields.
To enter a range of ports,
• enter the port number at the beginning of the range in the Sta
rt Port field.
• enter the port number at the end of the range in the End Port fiel
d.
DTMF Mode Control how the Router handles the tones that your telephone makes when you push
its bu
ttons. You should use the same mode your VoIP service provider uses.
RFC2833 - send the
D
TMF tones in RTP packets.
Inband - send the D
TMF 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.726) can distort the tones.
SIPInfo
- send the DTMF tones in SIP messages.
Tr
ansport Type The transport layer protocol used for SIP is UDP.
FAX Option This field controls how the Router handles fax messages.
G711 Fax
Pa
s
sthrough
Select this if the Router should use G.711 to send fax messages. The peer devices
must also use G.711.
Table 61 SIP > SIP Service Provider: Edit (continued)
LABEL DESCRIPTION