Axis M3037-PVE User Manual page 41

AXIS M3037-PVE Fixed Dome Network Camera
Voice over IP (VoIP)
Transport Settings
• Enable SIPS – Select to use Secure Session Initiation Protocol (SIPS). SIPS uses the TLS transport mode to encrypt traffic. If
you enable SIPS, you cannot select any other transport mode than TLS.
• Transport mode – Select the SIP transport mode for the account: UDP, TCP, or TLS.
• Allow port update messages through MWI – Message waiting indicator (MWI) notifies the user of changes in the
port settings.
The difference between SIPS (Enable SIPS) and SIP over TLS (Transport mode – TLS) is that SIPS ensures that each message transfer
is encrypted, while TLS only ensures encryption of the SIP traffic to the next node in the network.
SIP over UDP Transport mode – UDP is generally faster as the message will be sent without the handshakes that SIPS, SIP over TLS,
and SIP over TCP Transport mode – TCP offer.
Proxy Settings
A SIP proxy manages registration and routing requests from calling devices. The SIP proxy communicates with the private branch
exchange (PBX) in order to find a route that a call has to take to reach a device that is set in a different location or site.
Address - Enter the SIP proxy server's address.
Username - Enter a user name for the SIP proxy server if required.
Password - Enter a password for the SIP proxy server if required.
Make Test Calls
To make sure that calls can be made from the Axis product, you can make a test call:
1. Go to Applications > VoIP Client and click Configure under Account Settings.
2. From the list on the Account Settings page, select the account to make the test call from.
3. In the test call field, enter a valid SIP address to the other device. Use the format sip:<extension>@<domain> or
sips:<extension>@<domain>. For more information and examples, see About SIP Addresses on page 41.
4. Click Test call. For more information, see Transport Settings on page 41.
The call status is displayed. For more information, see Call Status on page 42.
5. To end the call, click End call.
About SIP Addresses
SIP addresses are used to identify users within a network just like you would use a phone number or an email address to contact a
friend or colleague.
Like email addresses, SIP addresses are a type of uniform resource identifier (URI) that include two user-specific parts, a user ID or
extension and a domain or IP address. Together with a prefix and the @ symbol, they make up a unique address. For example, if
Caesar of ancient Rome had both an email address and a SIP address, they would be mailto:caesar@ancientrome.it
and sip:caesar@ancientrome.it respectively. For local peer-to-peer calls only the IP address is required. For more
examples and descriptions, see below.
SIP addresses are also known as SIP URIs or, in some cases, SIP numbers. For more information, see the PBX's and service provider's
instructions.
Example
sip:192.168.0.90
sip:3468@172.25.33.142
sip:3468@voipprovider.com
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