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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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