SIP - Session Initiation Protocol

Telesis Systems Offering SIP

• Telesis Business Phone Systems:
  • PX24U Hybrid IP PBX Business Phone System
  • PX24M Hybrid IP PBX Business Phone System
  • PX24X Hybrid IP PBX Business Phone System
• Telesis Switching Systems:
  • X1 Large Capacity TDM - IP Telephony Switch
• Combined VoIP Gateway and Signaling Converters:
  • Stillink 200
  • Stillink 800
  • Stillink 3200

Standards

RFC 3261 SIP: Session Initiation Protocol.

Applications

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

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

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 

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.

Audio Codecs

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:

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

Echo Cancellation

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.

Call Routing

A Telesis system routes a call from the TDM network to the IP network according to:

• Dialed digits (called number)
• DPC
• Calling party number and information elements whenever available, such that:
  • Category of calling party
  • NOA of calling party
  • Numbering plan of calling party
  • Presentation status of calling party
  • Screening status of calling party