All calls between internal subscribers and subscribers who can be reached over external lines are handled as SIP calls by the R&S®NC VoIP router—even if the connection is between two ISDN subscribers.
The call router in the R&S®NC VoIP router handles the switching of the calls. The switching relies mainly on the information in two tables:
- For telephone numbers arriving at the call router, rules in the call-routing table are able to alter these numbers if needed and can decide which line to use for a call.
- The table for the locally registered user provides information about which terminal device is available at which internal telephone number.
The bandwidth reservation, QoS settings and firewall settings that are necessary for reliable transmission of voice data are carried out automatically by the R&S®NC VoIP router.
- When establishing a connection, the R&S®NC VoIP router checks (under consideration of the permitted
codecs) the maximum bandwidth that will be required.
- This bandwidth is then automatically reserved by the QoS module upon initiation of the connection.
- If negotiation shows that the maximum bandwidth is not available, the connection will not be made.
- If negotiations between the terminal devices can agree upon a codec with lower bandwidth requirements, then the reserved bandwidth will be reduced accordingly.
- All packets from ISDN users are given a DiffServ marking by the R&S®NC VoIP router (with SIP users,
the QoS marking is usually handled by the telephones or softphones):
- SIP packets for signaling are marked as CS1.
- RTP packets are marked as EF.
- The ports required for the transmissions are activated automatically.
