The basic features
The service is based on signaling protocols SIP and H.323. Routing is performed at central elements with high call capacity. Most institutions access the service using own voice gateway (in majority Cisco) connected through ISDN PRI (E1) interface to the classic phone exchange of the institution. These gateways are connected to interconnecting elements of the e-infrastructure. Another option for the institutions with own IP phone network is to agree upon the interface (interconnection points, protocols, contacts etc.).
The requirement is a quality network connection (ideally low latency and jitter and minimal packet loss) with capacity of 80 kbps per one call (for G.711, G.722 codecs), use of pre-defined interface, and maintaining contact and technical information up-to-date. The collaboration with technical contacts during solving problems related to the user equipment is a must (access to gateways).