SIP Advanced Configuration

This chapter covers some advanced aspects of SIP configuration and troubleshooting.

Gateway Clustering Support for SIP

This section covers gateway clustering support.

Synchronizing SIP Connections

SIP calls can be made across a ClusterXL gateway cluster or a third-party gateway cluster. For ClusterXL and third party gateway clusters (and when SIP connections must be synchronized across gateways): make sure that the Synchronize connections on Cluster option is selected. Select the option for all services used in rules that secure SIP connections through the gateway cluster.

To make sure SIP connections through a gateway cluster are synchronized:

1. In the SmartDashboard objects tree, select the Services tab.
2. Edit the SIP service that is used in rules that secure SIP connections through the gateway cluster.
3. In the Service Properties window of the SIP service, click Advanced.
4. Select Synchronize connections on Cluster.

   Note - The Synchronize connections on Cluster option is enabled by default

5. Click OK.
6. Install the policy.

Load Sharing of SIP Connections

SIP calls can be made across a ClusterXL gateway in High Availability mode or Load Sharing mode. In Load Sharing Mode, the Sticky Decision Function must be enabled. For more on the Sticky Decision Function, see the R76 ClusterXL Administration Guide.

Configuring SIP-T Support

To configure support for RFC 3372 Session Initiation Protocol for Telephones (SIP-T):

1. Add the $FWDIR/lib/user.def line on the Security Management server:

   sipt_hosts = { < first_ip, second_ip> , < first_ip, second_ip> , .... ....,< first_ip, second_ip> } ;

   Where first_ip and second_ip are the IP addresses between which (bi-directional) SIP-T are allowed. For example, to allow SIP-T between 192.1.1.1 and 192.1.1.2, and between 192.1.1.1 and 192.1.1.3 add this line:

   sipt_hosts = { < 192.1.1.1, 192.1.1.2> , < 192.1.1.1, 192.1.1.3> } ;

   If the file does not exist, create it.

2. Save the file.
3. Install the security policy.

Troubleshooting SIP

To get real-time information on SIP calls:

Run the fw tab -t sip_state -f command. This output is displayed:

• Control connection (source, destination).
• RTP connection (endpoint IP addresses).
• Call state (Initial, Call_Established, Call_Terminated...)
• Media type (audio, video, audio/video, application).
• Number of re-INVITE transactions used to implement VoIP features (such as Call Hold, Conference Call...)