Axis C8033 User Manual page 17

AXIS C8033 Network Audio Bridge
Voice over IP (VoIP)
• Secondary SIP Server – The Axis product will try to register on the secondary SIP server if registration on the primary SIP
server fails.
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. By default, TLS is used when media
encryption is activated.
• Media encryption – Encrypts media (audio and video) in SIP calls.
- SRTP Best Effort – Supports both encrypted and unencrypted media. Encryption is always used if available.
Supported crypto suites are AES_CM_128_HMAC_SHA1_80 and AES_CM_128_HMAC_SHA1_32.
- SRTP Mandatory – Supports only encypted media, that is, SIP calls are only set up if the remote
party offers SRTP media encryption. Supported crypto suites are AES_CM_128_HMAC_SHA1_80 and
AES_CM_128_HMAC_SHA1_32.
Note
If more than one TLS peer-to-peer account is created and they are using different media encryption, the strictest value is
applied to all of them. Registered accounts are not affected.
• 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 VoIP > 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 18.
4. Click Test call. For more information, see Transport Settings on page 17.
The call status is displayed. For more information, see Call Status on page 18.
5. To end the call, click End call.
Available from A1 Security Cameras
www.a1securitycameras.com email: sales@a1securitycameras.com
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