SIP Overview
SIP (Session Initialization Protocol) is an application-layer control (signaling) protocol used
on the V7111 for creating, modifying, and terminating sessions with one or more participants.
These sessions can include Internet telephone calls, media announcements and
conferences.
SIP invitations are used to create sessions and carry session descriptions that enable
participants to agree on a set of compatible media types. SIP uses elements called Proxy
servers to help route requests to the user's current location, authenticate and authorize users
for services, implement provider call-routing policies and provide features to users.
SIP also provides a registration function that enables users to upload their current locations
for use by Proxy servers. SIP, on the V7111, complies with the IETF (Internet Engineering
Task Force) RFC 3261 (see http://www.ietf.org).
V7111 Features
This section provides a high-level overview of some of the many V7111 supported features.
General Features
Superior, high quality Voice, Data and fax over IP networks.
Toll quality voice compression.
Enhanced capabilities including MWI, long haul, metering, CID and out door protection.
Proven integration with leading PBXs, IP-PBXs, Softswitches and SIP servers.
Spans a range of 2 to 24 FXS/FXO analog ports.
Selectable G.711 or multiple Low Bit Rate (LBR) coders per channel.
T.38 fax with superior performance (handling a round-trip delay of up to nine seconds).
Echo Canceler, Jitter Buffer, Voice Activity Detection (VAD) and Comfort Noise
Generation (CNG) support.
Comprehensive support for supplementary services.
Web Management for easy configuration and installation.
EMS for comprehensive management operations (FCAPS).
Simple Network Management Protocol (SNMP) and Syslog support.
SMDI support for Voice Mail applications.
ThroughPacket™ proprietary feature that aggregates payloads from several channels
into a single IP packet to reduce bandwidth overhead.
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3Com
VCX V7111 VoIP Gateway User Guide
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