Introduction
There
are many applications of the Internet that require
the creation and management of a session, where
a session is considered an exchange of data between
associations of participants. The implementation
of these applications is complicated by the practices
of participants: users may move between endpoints,
they may be addressable by multiple names, and
they may communicate in several different media
– sometimes simultaneously. Numerous protocols
have been authored that carries various forms
of real-time multimedia session date such as voice,
video, or text messages. The Session Initiation
Protocol (SIP) works in concert with these protocols
by enabling Internet endpoints (called user agents)
to discover one another and to agree on a characterization
of a session they would like to share. For locating
a prospective session participants, and for other
functions, SIP enables the creation of an infrastructure
of network hosts (called proxy servers) to which
user agents can send registrations, invitations
to sessions, and other requests. SIP is an agile,
general-purpose tool for creating, modifying,
and terminating sessions that works independently
of underlying transport protocols and without
dependency on the type of sessions that is being
established.
Session Initiation Protocol
(SIP)
SIP is can application-layer control protocol
that can establish, modify, and terminate multimedia sessions
(conferences) such as Internet telephony calls. SIP can also
invite participants to already existing sessions, such as
multicast conferences. Media can be added to (and removed from)
an existing session. SIP transparently supports name mapping
and redirections services, which supports personal mobility –
users can maintain a single externally visible identifier
regardless of their network location.
Features & Benefits:
-
Exceptional call qualities,
both locally as well as internationally.
-
Able to make calls
to local, Mobile, STD, IDD, 1-300, 1-800
as well as emergency numbers (e.g. 999,
994 etc).
-
Able to receive calls
from any PSTN as well as Mobile; as users
are provided with a DID number (e.g. 03-2782XXXX).
-
Enjoy great savings
making outstation as well as IDD calls,
while calls within the
myVoiceLynx network are
FREE.
-
Value Added Services
such as voicemail, video call, call forwarding,
SMS, call conference, music ring back
tone etc.
-
Optional access to
voicemail via e-mail.
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