Archive for the ‘VOIP’ Category

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SIP Trunking


2010
05.18

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Posted in INFRASTRUCTURE, LINUX, VOIP | No Comments »

A SIP Trunk is a service; telephone call is routed over the IP backbone of a carrier using VoIP technology. SIP Trunks are used in conjunction with an IP-PBX and are thought of as replacements for traditional PRI or analog circuits. The popularity of SIP Trunks is due primarily to the cost savings of SIP, along with the increased reliability as backed by the SLAs of SIP Trunk Providers.

A SIP trunk is a service offered by an ITSP (Internet Telephony Service Provider) that permits businesses that have a PBX installed to use Voice-over-IP (VoIP) also outside the enterprise network by using the same connection as the Internet connection.

SIP trunking is a way to enjoy significant savings on your current phone bill. Using an Internet connection right from your current PBX, a SIP trunk uses SIP (Session Initiation Protocol) for a VoIP connection.

In these tougher economic times, more and more businesses, both small and large at looking at the advantages of SIP trunking and creating call treatments in their PBXs to take more chargeable calls out through these cost saving routes. And this trend is likely to substantially increase over years.

SIP trunking takes data, voice and video out over your Internet connection.

Some of the benefits of SIP trunking include:
• Much lower rates on long distance calls, International calls and in-bound toll free calls.
• The ability to establish virtual numbers in other geographic areas so that callers can use these numbers instead on more costly 800 numbers.
• Using virtual numbers from other areas to establish a “point of presence” for the company.
• Reduced costs as no TDM cards are required.
• Getting the benefits of Hosted VoIP without having to abandon existing equipment or investing in IP phones.
• The ability to easily add more calling trunks without the need for expensive digital PRI / analog station (port) cards.
• The ability to use an IAD for connecting regular analog phones, instead of having to buy more expensive equipment.
• Expansion of lines is dependent on bandwidth, which can easily be increased if needed. Additional lines can be set for compressed codecs (G729 and others) to use less bandwidth per call.

Ref: TMCnet.com

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IP Phone for Telecommuters and Remote Workers


2010
04.22

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Posted in INFRASTRUCTURE, Network, VOIP | No Comments »

Employees Working from Home, Temporary or Emergency Locations Can Easily Install IP Desktop Phones Connected Securely with VPN via the Internet. Improved Business Continuity, Productivity and Cost-Savings from Distributed Enterprise Networks.

Telecommuter solution that embeds virtual private network (VPN) remote capabilities into IP telephones. Businesses can use this solution to more easily and securely extend headquarters-quality communications to employees working from any home office, temporary work site or emergency location.

The number of telecommuters and home-based workers are expected to continue rising over the next few years, according to IDC Research. By 2009, 10 million U.S. workers, or 10% of its workforce, will be telecommuting. In Western Europe, 9 million workers are expected to telecommute, representing a growth rate of 10% from 2004-2009¹.

Ref: Cisco & Avaya

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SIP: Session Initiation Protocol


2010
04.01

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Posted in INFRASTRUCTURE, Network, VOIP | No Comments »

SIP (Session Initiation Protocol) is an application-layer control protocol that can establish, modify, and terminate multimedia sessions such as Internet telephony calls (VOIP). 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 redirection services, which supports personal mobility – users can maintain a single externally visible identifier regardless of their network location.

SIP supports five facets of establishing and terminating multimedia communications:
• User location: determination of the end system to be used for communication;
• User availability: determination of the willingness of the called party to engage in communications;
• User capabilities: determination of the media and media parameters to be used;
• Session setup: “ringing”, establishment of session parameters at both called and calling party;
• Session management: including transfer and termination of sessions, modifying session parameters, and invoking services.

SIP is a component that can be used with other IETF protocols to build a complete multimedia architecture, such as the Real-time Transport Protocol (RTP ) for transporting real-time data and providing QoS feedback, the Real-Time streaming protocol (RTSP ) for controlling delivery of streaming media, the Media Gateway Control Protocol (MEGACO) for controlling gateways to the Public Switched Telephone Network (PSTN), and the Session Description Protocol (SDP ) for describing multimedia sessions. Therefore, SIP should be used in conjunction with other protocols in order to provide complete services to the users. However, the basic functionality and operation of SIP does not depend on any of these protocols.

SIP provides a suite of security services, which include denial-of-service prevention, authentication (both user to user and proxy to user), integrity protection, and encryption and privacy services.

SIP works with both IPv4 and IPv6. For Internet telephony sessions, SIP works as follows:-

Callers and callers are identified by SIP addresses. When making a SIP call, a caller first locates the appropriate server and then sends a SIP request. The most common SIP operation is the invitation. Instead of directly reaching the intended caller, a SIP request may be redirected or may trigger a chain of new SIP requests by proxies. Users can register their location(s) with SIP servers. SIP addresses (URL) can be embedded in Web pages and therefore can be integrated as part of powerful implementations such as Click to talk.