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StillinkX100SessionInitiationProtocol

Capabilities

The applied standard in Stillink x100 systems is RFC 3261.

Stillink x100 systems can register to multiple SIP registrars/proxies/servers at the same time. This allows address resolution of a Stillink x100 system from either side and results in flexibility for multipath VoIP access applications.

Stillink x100 systems support numerous SIP users (entities) which can be user agents or registrars/proxies/servers. Number/IP translation is performed through advanced routing algorithms. Together with the integrated registrar/proxy/server, call authorization, call management, enhanced billing functions, flexible routing algorithms, and extensive business telephony features make a Stillink x100 system serve as a feature-rich communication platform.

Connecting to the long distance call operator can be both over TDM and IP at the same time. The Stillink x100 system may register at the external registrar/proxy/server of the operator as an option. With the advanced routing algorithms and alternate routing capability, TDM calls from a terminal equipment connected to the system may be routed to a selected operator over the IP or PSTN. Alternate routing capability provides automatic fall back to the PSTN if the IP network is inaccessible.

Furthermore, it is possible to define which release causes stated in SIP response codes will lead to the alternate routes being used.

Physical Layer

The physical layer for IP in Stillink x100 systems is 10-100 BaseT Ethernet. Several ethernet and other properties for IP Telephony are programmable.

Audio Codecs

Stillink x100 systems are equipped with well-known audio codecs featuring audio compression as well. Audio codec preference list and properties such as silence suppression (VAD-Voice Activity Detection), frame length are programmable for the system. Currently available codecs for VoIP calls are:

  • G.711 (A and u)
  • G.723.1 (5.3kbps, 6.4kbps)
  • G.729
  • G.729AB

Echo Cancellation

In Telesis systems, an AT&T certified G.168 echo canceler meets and exceeds G.168-2002 standards. The echo canceler can operate with delays as high as 128msec. It is better than industry standard cancelers under the most important and difficult conditions like double-talk and the presence of background noise.

Fax Pass-through Over SIP

It is a method for the fax transmission over SIP. In this method, the fax transmission is similar to a G.711 -based VoIP voice call after T.30 fax tones detection. On detection of fax tones, the used VoIP codec is switched to G.711.

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Page last modified on January 20, 2020, at 06:42 AM