Telesis STILLINK: Combined VoIP Gateway and Signaling Converters
SIP - Session Initiation Protocol
HOW TO BUY
BECOME OUR RESELLER / DEALER
SIP - Session Initiation Protocol
Telesis Systems Offering SIP
RFC 3261 SIP: Session Initiation Protocol.
Telesis systems integrate both packet and circuit switching technology. A Telesis system featuring an integrated SIP registrar/proxy/server provides an economical way for administrators to manage a central database of phone numbers without the expense of a separate-box registrar/proxy/server solution.
The integrated SIP registrar/proxy/server may serve to numerous SIP user agents
Capability of registering to multiple SIP registrars/proxies/servers at the same time provides multipath VoIP access
Telesis systems support numerous SIP users (entities) which can be user agents or registrars/proxies/servers. Number/IP translation is performed through an advanced routing algorithm. Together with the integrated registrar/proxy/server, call authorization, call management, enhanced billing functions, flexible routing algorithms, and extensive business telephony features make a Telesis system serve as a feature-rich communication platform.
SIP entities may be user agents and registrars/proxies/servers at the same time
Connecting to the long distance call operator can be both over TDM and IP at the same time. The Telesis 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 unaccessible.
Advanced routing algorithms and alternate routing capability ensure call progress
Another SIP related feature powering Telesis systems is the embedded WepPhone service. Telesis WebPhone is a SIP softphone or client applet hosted in Telesis IP PBX, Telesis TDM - IP telephony systems, Stillink access gateways and Stillink signaling converters. The applet can be run from any Java enabled web browser. Since it is a standard Java applet, no software installation is required or no plugin is installed. The applet is just loaded temporarily onto the client`s browser. It can be used wherever is an access to the Internet. A web browser, a microphone, and a headset are sufficient to make calls.
Integrated Web Server of the Telesis System Hosts the Java SIP Client Applet
Layers and Options
The physical layer of SIP protocol in Telesis systems is 10/100 BaseT Ethernet. Several ethernet and other properties for SIP are programmable.
Telesis 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:
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.
Route and Routeset Configuration
In a Telesis system, a SIP user agent may have its own route number. It is possible to define numerous distinct routes. A given route to a particular destination and its accompanying alternate routes are grouped in a routeset. Each route in a routeset has a priority order. Alternate routes may be SIP user agents or TDM (PSTN) lines. Routing to the next priority alternate route is possible in the event that a route becomes unavailable.
A Telesis system routes a call from the TDM network to the IP network according to:
We recommend you to visit Telesis Wiki Pages for technical readings.
Copyright Telesis A.S. 2006-2013