AXIS C1610-VE Network Sound Projector
Additional settings
Additional settings
Calibrate and run a remote speaker test
You can run a speaker test to verify from a remote location that a speaker is working as intended. The speaker performs the test by
playing a series of test tones that are registered by a microphone.
Note
The test must be calibrated from its mounted position at the installation site. If the speaker is moved or if its local
surroundings change, for instance if a wall is built or removed, the speaker should be re-calibrated
If you calibrate the speaker when an external microphone is connected, then the external microphone will be used for the
calibration. This means that whenever you run a speaker test in the future, the external microphone must be in the exact
same position and set to the same gain, as it did during the initial calibration. You can avoid this by disconnecting the
external microphone when you calibrate and test your speakers, since the internal microphone will be used instead.
During calibration, it is recommended that someone is physically present at the installation site to listen to the test tones and
ensure that the test tones are not muffled or blocked by any unintended obstructions in the speaker’s acoustic path.
1. Go to the device interface >
Audio > Speaker test
.
2. To calibrate the audio device, click
Calibrate
.
Note
Once the Axis product is calibrated, the speaker test can be run at any time.
3. To run the speaker test, click
Run the test
.
Note
It is also possible to run the calibration by pressing the control button on the physical device. See
to identify the control button.
Set up direct SIP (P2P)
Use peer-to-peer when the communication is between a few user agents within the same IP network and there is no need for extra
features that a PBX-server could provide. To better understand how P2P works, see
Peer-to-peer SIP (P2PSIP) on page 12
.
For more information about setting options, see
.
1. Go to
System
>
SIP
>
SIP settings
and select
Enable SIP
.
2. To allow the device to receive incoming calls, select
Allow incoming calls
.
3. Under
Call handling
, set the timeout and duration for the call.
4. Under
Ports
, enter the port numbers.
-
SIP port
– The network port used for SIP communication. The signaling traffic through this port is non-encrypted.
The default port number is 5060. Enter a different port number if required.
-
TLS port
– The network port used for encrypted SIP communication. The signaling traffic through this port is
encrypted with Transport Layer Security (TLS). The default port number is 5061. Enter a different port number
if required.
-
RTP start port
– Enter the port used for the first RTP media stream in a SIP call. The default start port for
media transport is 4000. Some firewalls might block RTP traffic on certain port numbers. A port number must
be between 1024 and 65535.
5. Under
NAT traversal
, select the protocols you want to enable for NAT traversal.
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