- SIP standard according to RFC 3261
- RTP / RTCP according to RFC 1881
- Up to 25 SIP accounts configurable (from Rel. 7.3x)
- Support of up to 16 IP- system telephones
- TAPI Support for IP - system telephones
- Support of up to 30 IP terminals
- Telephony functions, such as: call waiting, call back, alternation between lines, 3-party conferences, call transfer with and without announcement, call forwarding, suffix dialling
- Voice compression (codecs) according to G.711, G.723.1, G.726, and G.729a/b
- Codec preselection
- DTMF inband / outband
- The module VoIP - VPN gateway enables the transition of IP voice data into the traditional voice network and vice versa.
oIP phones or IP systems telephones of the PBX can be integrated into the traditional voice network (ISDN or POTS). - Traditional end devices (analogue, ISDN) of the PBX are able to use Internet telephony (VoIP).
- Quality of Service: measures for quality assurance for VoIP
- Prioritisation of RTP packets: RTP packets coming from the logged on IP end devices are treated with higher priority by the gateway (DSCP header configurable).
- Bandwidth management: reservation of bandwidth for VoIP connections
- Echo compensation: local echo compensation to improve the voice understandability (max. 16 ms)
Features module VoIP-VPN Gateway