Filter
Top Products
Global SIP Settings
41-001160-03, Rev 00, Releaes 2.4 4-101
Configuring Network and Session Initiation Protocol (SIP) Features
2 Enter an RTP Port Base in the RTP Port field. Default is 3000.
The RTP Port indicates the port through which the RTP packets are sent. This value must specify the
beginning of the RTP port range on the gateway or router. The RTP port is used for sending DTMF
tones and for the audio stream. Your network administrator may close some ports for security
reasons. You may want to use this parameter to send RTP data using a different port.
Note: The phones support decoding and playing out DTMF tones sent in SIP INFO requests. The
following DTMF tones are supported:
• Support signals 0-9, #, *
• Support durations up to 5 seconds
3 Enable the "Basic Codecs (G.711 u-Law, G.711 a-Law, G.729)" field by checking the check box.
(Disable this field by unchecking the box. Default is disabled).
Enabling this parameter allows the IP phone to use the basic Codecs when sending/receiving RTP
packets.
4The "Force RFC2833 Out-of-Band DTMF" field is enabled by default. Disable this field by
unchecking the box.
Enabling this parameter forces the IP phone to use out-of-band DTMF according to RFC2833.
5 Enter a "Customized Codec Preference List". For example,
payload=8;ptime=10;
silsupp=on,
payload=0;ptime=10;
silsupp=off
Valid values are:
For this parameter, you specify a customized codec list which allows you to use the preferred Codecs
for this IP phone. Default for the “Customized Codec Preference List” is blank.
Aastra Web UI
Step Action
Attribute Value
payload 0 for G.711 u-Law
8 for G.711 a-Law
18 for G.729a
ptime (in milliseconds) 5, 10, 15, 20.....90
silsupp on
off
Draft 1