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41-001343-02 REV04 – 05.2014 3-42
RTP Settings
You can configure the following RTP settings.
Whitelist Proxy sip whitelist This parameter enables/disables the whitelist proxy feature, as follows:
• Set to 0 to disable the feature.
• Set to 1 to enable the feature. When this feature is enabled, an IP phone
accepts call requests from a trusted proxy server only. The IP phone
rejects any call requests from an untrusted proxy server.
For more information about Whitelist Proxy see Chapter 6, the section,
“Whitelist Proxy” on page6-15.
XML SIP Notify sip xml notify event Enables or disables the phone to accept or reject an aastra-xml SIP NOTIFY
message.
Note:
To ensure the SIP NOTIFY is coming from a trusted source, it is recom-
mended that you enable the Whitelist feature (Whitelist Proxy parameter)
on the IP phone. If enabled, and the phone receives a SIP NOTIFY from a
server that is NOT on the whitelist (i.e. untrusted server), the phone rejects
the message.
For more information about XML SIP Notify see Chapter 6, the section,
“XML SIP Notify Events” on page5-256.
Parameter In
IP Phone UI
Parameter in
Aastra Web UI
Parameters in Configuration
Files
Description
RTP Port Base RTP Port sip rtp port 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. Default is 3000.
For more information, see Chapter 4, “Real-time Transport Pro-
tocol (RTP) Settings” on page4-74.
N/A Force RFC2833 Out of
Band DTMF
sip out-of-band dtmf Enables or disables out-of-band DTMF. Enabling this parame-
ter forces the IP phone to use out-of-band DTMF according to
RFC2833. Valid values are 0 (disabled) and 1 (enabled). Default
is 1 (enabled).
For more information, see Chapter 4, “Real-time Transport Pro-
tocol (RTP) Settings” on page4-74.
N/A DTMF Method
(Global and Per-Line)
sip dtmf method
(global)
sip lineN dtmf method
(per-line)
Sets the dual-tone multi-frequency (DTMF) method to use on
the IP phone on a global or per-line basis. Valid values are 0
(RTP), 1 (SIP INFO), or 2 (BOTH). Default is 0 (RTP).
For more information, see Chapter 4, “Real-time Transport Pro-
tocol (RTP) Settings” on page4-74.
N/A RTP Encryption
(Global and Per-Line)
sip srtp mode
(global)
sip lineN srtp mode
(per-line)
This parameter determines if SRTP is enabled on this IP phone,
as follows:
• If set to 0, then disable SRTP.
• If set to 1 then SRTP calls are preferred.
• If set to 2, then SRTP calls only are generated/accepted.
For more information, see Chapter 4, “Real-time Transport Pro-
tocol (RTP) Settings” on page4-74.
Parameter in
Aastra Web UI
Parameters in Configuration Files Description