SIP Softphone(s)

Many telephone systems (PBXs) make the operation of telephones possible that have been implemented according to the SIP standard. estos UCServer supports the central integration of such telephone systems. This integration allows estos ProCall client users to use their PCs as softphones in order to make telephone calls through the telephone system. To do this, the ProCall client gets one or more lines, that respectively correspond to one telephone from UCServer.

To configure UCServer, one such line that is respectively responsible for registration of a certain telephone number must first be added. Afterwards, the line will be assigned to a user with the help of this telephone number. Thereby, the ProCall client users can use the telephone system for making telephone calls through UCServer.

UCServer already has the SIP modules necessary for PBX integration, which assume control of the call signals. In addition, UCServer contains a media server that binds the PBX on the one hand and the ProCall clients on the other hand with each other. By using the media server, the media streams will respectively be converted into the correct format. On the client side, the media streams are encrypted (DTLS-SRTP), even when the PBX does not provide encryption. The telephonic accessibility of the users located on the Internet is another job of the media server. If a ProCall Mobile client is outside of the reach of the internal WLAN or, for example, a PC client is in a home office, the central PBX can continue to be used for making telephone calls.


Technical Information

The telephone system must allow registrations through a LAN interface in accordance with the SIP standard (RFC 3261). UCServer does not need a SIP-specific license. However, some telephone systems need licenses in order to register SIP softphones with the telephone system.

The media server provides the G.711 (PCMU, PCMA) audio codecs in the direction of the PBX. In the direction of the ProCall clients, Opus is generally used. This also provides good audio quality using little LAN/WAN bandwidth. By encrypting according to the DTLS/SRTP procedure, the media server uses the highest security standard that is currently normal in VoIP products.

Accessibility on the Internet through the media server is achieved through the use of TURN & STUN server services. If TURN & STUN servers have not been configured, communication within the LAN (local area network) will be possible.


Version 6.0