Goal
This scenario is targeting the setting up of basic voice services for an institution. We are proposing a solution where the IP PBX is complementary to the PBX of the institution.
The features that we will be explored are:
- authenticated login
- user provisioning
- gateway to the PSTN
- gateway between SIP and ?.323
- local dial plan
- encryption (endpoint-to-endpoint)
- codec support
- handling of both numeric and alpha-numeric addresses
- accounting
- qos
- availability checks
Applicability
- SIP interface with SIP proxy (OpenSER)
- H.323 interface with H.323 gatekeeper (GnuGk)
- SIP to H.323 gateway (Asterisk)
- Radius authentication (FreeRadius, Radiator)
- LDAP identity management infrastructure (OpenLDAP)
- accounting (MySQL)
Discussion of the interconnection of the components
Prerequisites (OS, dependencies on other software)
- a PBX (to interface to local institution phones)
- a gateway (to connect from the IP world to the PBX)
- authentication infrastructure (Radius or LDAP)
Configuration (OS agnostic)
OpenSER - The SIP interface
Sample config for a typical OpenSER setup with:
- authentication
- multi-domain handling
- protected from open-relay calls
- loose routing
- ENUM routing
- SIP.edu incoming call resolution
- NAT support
PSTN Gateway - The gateway to the PBX
Cisco Gw
Cisco CallManager
PRI card on a PC with Asterisk
GnuGk
Sample config for interfacing with the H.323 world:
- authentication
- ENUM routing