Axis C1410, Руководство пользователя, Страница: 8 / 40

AXIS C1410 Network Mini Speaker
Additional settings
- 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.
Note
Use NAT traversal when the device is connected to the network from behind a NAT router or a firewall. For more information
see
NAT traversal on page 12 .
6. Under Audio , select at least one audio codec with the desired audio quality for SIP calls. Drag-and-drop to change
the priority.
7. Under Additional , select additional options.
- UDP-to-TCP switching – Select to allow calls to switch transport protocols from UDP (User Datagram Protocol)
to TCP (Transmission Control Protocol) temporarily. The reason for switching is to avoid fragmentation, and
the switch can take place if a request is within 200 bytes of the maximum transmission unit (MTU) or larger
than 1300 bytes.
- Allow via rewrite – Select to send the local IP address instead of the router's public IP address.
- Allow contact rewrite – Select to send the local IP address instead of the router's public IP address.
- Register with server every – Set how often you want the device to register with the SIP server for the existing
SIP accounts.
- DTMF payload type – Changes the default payload type for DTMF.
8. Click Save .
Set up SIP through a server (PBX)
Use a PBX-server when the communication should be between an infinite number of user agents within and outside the IP network.
Additional features could be added to the setup depending on the PBX-provider. To better understand how P2P works, see
Private
Branch Exchange (PBX) on page 11 .
For more information about setting options, see SIP on page 26 .
1. Request the following information from your PBX provider:
- User ID
- Domain
- Password
- Authentication ID
- Caller ID
- Registrar
- RTP start port
2. To add a new account, go to System > SIP > SIP accounts and click + Account .
3. Enter the details you received from your PBX provider.
4. Select Registered .
5. Select a transport mode.
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