The SBC can be configured for special cases, e.g. Media Anchoring and Auto IP, XML Bodies of SIP Messages, TLS/SRTP, Record Route, SIP Options Request, Emergency Calls, Call Admission Control and Firewalls/SIP ALGs.
The following SBC (Session Border Controller) special cases can be configured:
Lotus Sametime Unified Telephony Geo-Separated NodesParent topic: SBC (Session Border Controller)
When an SBC is deployed in the branch (e.g. Branch SBC) and a centralized SBC is also deployed, it is called a cascaded SBC scenario.
In this scenario the branch SBC performs port mapping and therefore it is not necessary to do this again in the centralized SBC.
The trunk-port-per-endpoint value should not be used in the alter-contact parameter of the registration plans of these branches.
Media Anchoring and Auto IP
XML Bodies of SIP Messages
Core Side TCP (Record-Route headers)
SIP Options Request
Emergency Calls (E-911)
The Lotus Sametime Unified Telephony emergency call solution uses a callers IP address or subnet in order to associate a LIN (Location Information Number) with the emergency call.
When users connect to Lotus Sametime Unified Telephony via an SBC the topology hiding features of the SBC hide the users IP address and subnet (on the "outside" of he SBC) from Lotus Sametime Unified Telephony (on the "inside" of the SBC).
The SBC must be configured with separate SIP interfaces (on the "inside" of the SBC) for each subnet on the "outside" of the SBC. Registration and Dial Plans must be created for each subnet on the "outside" of the SBC in order to route the SIP messages to Lotus Sametime Unified Telephony via the associated interface on the "inside" of the SBC. This allows Lotus Sametime Unified Telephony to associate a LIN with the IP address of each SIP interface on "inside" of the SBC.
Call Admission Control
The Lotus Sametime Unified Telephony call admission control solution has similar considerations as the Lotus Sametime Unified Telephony emergency call solution. In order for Lotus Sametime Unified Telephony to apply bandwidth limitations to traffic to/from individual locations / subnets the SBC must be configured with separate SIP interfaces (on the "inside" of the SBC) for each subnet / geographical location on the "outside" of the SBC.
Firewalls/SIP ALGs (Application Level Gateways)
Some firewalls and routers, such as Cisco 2800 and 3600 series edge routers, have a built-in SIP traversal function, also known as a "SIP fix-up" mechanism. This capability is often referred to as a SIP ALG. The SIP ALG function in the firewall or router must be disabled when used together with a centralized SBC in the Lotus Sametime Unified Telephony environment.
DSCP/TOS values in IP packet headers
The SBC provides the capability to set DSCP values in the SIP signaling packets sent by the SBC. This capability is configured by the session-config/sip-settings inleg-tos and outleg-tos parameters. The default value for these setting is preserve and this is the correct value for standard Lotus Sametime Unified Telephony deployments, where the DSCP values are set in IP packets originated by the endpoints and the SBC is required to pass these values transparently.
How to Configure Lotus Sametime Unified Telephony Geo-Separated Nodes
How to Configure XML Bodies of SIP Messages
When Keysets are deployed behind an SBC (Session Border Controller) then the SBC is required to perform address translation on the remote-uri IP address within the keyset-info XML document that will be present in a SIP NOTIFY request for keyset-info events sent from the Lotus Sametime Unified Telephony to the Keyset SIP UA (User Agent).
How to Configure TLS/SRTP
How to Configure Route Headers
How to configure core side TCP (Record-Route headers)
For routes from SBC to Lotus Sametime Unified Telephony and vice versa which use TCP transport, the following parameters must be included in the appropriate session-configs to force the SBC to include a Record-Route header in requests sent to Lotus Sametime Unified Telephony. Lotus Sametime Unified Telephony needs the Record-Route header to correctly route in-dialog requests back to the SBC.
How to Configure SIP Options Request