What can I do here?
Use this window to create or edit a SIP Proxy (also known as a SIP server or Registrar).
Getting Here - Object Explorer > New > Network Object > More > VoIP Domain > SIP Proxy |
There are five types of VoIP Domain objects:
In many VoIP networks, the control signals follow a different route through the network than the media. This is the case when the call is managed by a signal routing device. Signal routing is done in SIP by the Redirect Server, Registrar, and/or Proxy. In SIP, signal routing is done by the Gatekeeper and/or gateway.
Enforcing signal routing locations is an important aspect of VoIP security. It is possible to specify the endpoints that the signal routing device is allowed to manage. This set of locations is called a VoIP Domain. For more information refer to the R80.30 VoIP Administration Guide.
Below is a list of supported SIP topologies. The table also lists NAT that you can configure with each topology. it with. SIP can use a Proxy (or Registrar). If there is more than one proxy device, signaling passes through one or more of them. After the call is set up, the media can pass from endpoint to endpoint directly, or through one or more of the proxies.
Deployment |
Supports No-NAT |
Supports NAT for Internal Phones - Hide/Static NAT |
Supports NAT for Proxy - Static NAT |
Description |
---|---|---|---|---|
SIP Endpoint to Endpoint |
Yes |
Static NAT only |
Not applicable |
|
SIP Proxy in External Network |
Yes |
Yes |
Not applicable |
|
SIP Proxy to SIP Proxy |
Yes |
Yes |
Yes |
|
SIP Proxy in DMZ |
Yes |
Yes |
Yes |
|
For complete information on NAT configuration, see the R80.30 Security Management Administration Guide.
Below are some exceptions when you use SIP with NAT:
You can use SIP with NAT with these exceptions: