Global SIP Settings
41-001343-02 REV05 – 07.2014
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When an active call is using SRTP (i.e. when an SRTP enabled IP phone initiates a call and the receiving phone is also SRTP
enabled) and the transport protocol is set to TLS, the IP Phone UI displays a “lock” icon, indicating that the call is secure. If
one of the phones does not support SRTP and/or TLS is not enabled, the IP Phone UIs do not display the lock icon, indicat-
ing that the call may not be secure.
You can configure SRTP on a global or per-line basis using the configuration files or the Aastra Web UI.
Silence Suppression
In IP telephony, silence on a line (lack of voice) uses up bandwidth when sending voice over a packet-switched system.
Silence suppression is encoding that starts and stops the times of silence in order to eliminate that wasted bandwidth.
Silence suppression is enabled by default on the IP phones. The phone negotiates whether or not to use silence suppres-
sion. Disabling this feature forces the phone to ignore any negotiated value.
You can configure silence suppression on a global-basis using the configuration files or the Aastra Web UI.
Option to Include/Remove Silence Suppression Attribute from SDP Offer
The parameter
sip remove silence suppression offer
is available allowing administrators the ability to control whether or
not the silence suppression attribute should be included in the Session Description Protocol (SDP) offer.
If enabled (1), the silence suppression attribute will be removed from the SDP offer. If disabled (0), the attribute will not be
removed from the SDP offer. This parameter is disabled by default and requires a reboot if the value of the parameter has
changed. You can configure this parameter using the configuration files only.
Configuring RTP Features
Use the following procedures to configure the RTP features on the IP phone.
Note:
If you enable SRTP, then you should also enable Transport Layer Security (TLS). This prevents capture of the key used for
SRTP encryption. To enable TLS, set the
Transport Protocol
parameter (located on the Global SIP Settings menu) to
TLS
.
Configuration Files
For specific parameters you can set for RTP features in the configuration files, see Appendix A, the section,
“RTP, Codec, DTMF Global Settings”
on
IP Phone UI
1.
Press
on the phone to enter the Options List.
2.
Select
Administrator Menu.
3.
Enter your Administrator password.
Note:
The IP Phones accept numeric passwords only.
4.
Select
SIP Settings.
5.
Select
RTP Port Base
to change the RTP port base setting. Default is
3000
.
6.
Press
Done
(2 times) to save the change.
Note:
The session prompts you to restart the IP phone to apply the configuration settings
7.
Select
Restart.
