These preconfigured SIP services are available for gateways of version R80.xx or higher.
Services |
Port |
Protocol Type |
Description |
---|---|---|---|
|
UDP 5060 |
|
This service enforces signal routing. Use a VoIP Domain in the source or destination of a rule, together with this service. When you use this service, registration messages are tracked and a database is maintained that includes the details of the IP phones and the users. If an incoming call is made to a Hide NATed address, the Security Gateway confirms the user exists in the SIP registration database. This can prevent DoS attacks. |
|
TCP 5060 |
|
Used for SIP over TCP. |
|
Not set |
Not set |
This service allows a SIP connection to be opened on a dynamic port and not on the SIP well-known port. |
|
TCP 5061 |
None |
Allows SIP over TLS to pass without inspection. It requires that you open the media ports manually. |
|
TCP 5061 |
|
SIP over non-encrypted TLS and authenticated only. NAT is not supported for connections of this type. |
These legacy SIP services are used for gateways of version R75.40 and below, if not enforcing handover. Do not use these services for R.80.xx (or higher).
Services |
Purpose |
---|---|
|
Use sip_any for VoIP equipment that uses SIP UDP. Do not place a VoIP Domain in the Source or Destination of a rule. Instead, use * Any or a Network Object, together with one of these services.
Note - If a VoIP Domain is used with this service, the packet is dropped. Important - Do not use this service in the same rule with the sip service because they contradict each other. |
|
Use sip-tcp_any for VoIP equipment that uses SIP TCP. Use this service if you do not enforce signal routing. In that case, do not place a VoIP Domain in the Source or Destination of a rule. Instead, use * Any or a Network Object together with the sip_any-tcp service. Note - If a VoIP Domain is used with this service, the packet is dropped. Important - Do not use this service in the same rule with the sip-tcp service because they contradict each other. |
Legacy Solution for SIP TLS Support
If you are not able to use the sip_tls_authentication service, add these two rules instead:
AND
This can happen if connections are encrypted by TLS, or NAT must be done on the connections.
Important - SIP signaling and data is not inspected if you open all high UDP ports. The connection is not-secured.
To configure support for SIP TLS in environments where a secure solution is not available:
The rule below shows that the phones send data directly to each other, and not through the proxy.
No |
Name |
Source |
Destination |
VPN |
Services & Applications |
Action |
Track |
---|---|---|---|---|---|---|---|
1 |
Transmit through proxy |
SIP Proxy SIP Phones |
SIP Phones SIP Proxy |
* Any |
TCP: sip_tls_not_inspected |
Accept |
Log |
2 |
Transmit directly |
SIP Phones |
SIP Phones |
* Any |
UDP: udp-high-ports |
Accept |
Log |
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 |
---|---|---|---|---|
Yes |
Static NAT only |
Not applicable |
|
|
Yes |
Yes |
Not applicable |
|
|
Yes |
Yes |
Yes |
|
|
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: