Interconnect Telecom is a leading international VoIP provider targeted for business clients with an expertly engineered IP network which offers a higher level of security, features and functions than the public internet. Companies are able to migrate to IP Communications from traditional TDM, using their reference CPE with a gateway device or all at once with an IP PBX. Such flexibility enables communication between any device which is IP allowed, including computers, SIP phones, smart phones and PDAs.
SIP Trunking uses Session Initiation Protocol (SIP) for providing voice communication between an Internet Telephony Service Provider (ITSP) as Interconnect Telecom and a company's IP-PBX Session Initiation Protocol (SIP). ITSPs use VoIP communication over the internet. ITSPs difference from ISPs (Internet Service Providers) is that voice communication is enabled with SIP and because only an internet connection is provided.
An Interconnect Telecom SIP Trunk provides coherent access between the internet and the PSTN resulting in cost savings, because calls travel most of the distance over the internet, thereby avoiding the high costs of the PSTN.
SIP Trunking makes a reasoned connection from one point to another through the internet, making it possible to process voice communication as just another application and deliver it as the piling of data. Before this change, voice signals had moved over physical wires. The aim is to have voice signal travelling on the worldwide PSTN (Public Switched Telephone Network) as quickly as possible. The reason is the costs of using the physical wires, which is much higher than the free internet.
If both the evoking and admitting sides of communication use VoIP (Voice over Internet Protocol) for voice communication, then it is possible to avoid PSTN charges altogether. If both parties use SIP to start voice communication, their calls can move onto the internet for no charge. The only charge is the rate charged by their ISP provider.
When companies use Interconnect Telecom SIP Trunking, calls mostly travel over the Interconnect Telecom IP Network instead of on the PSTN and then drop back down to the destination/facility in the last mile. If both participating sides use Interconnect Telecom SIP Trunking, in such a case the last mile connection to the PSTN is not necessary. The last mile is the physical line between the local phone company switching facility and the destination of the call. The less time a call spends on the PSTN, the lower the cost will be. Interconnect Telecom SIP Trunking services significantly lower local and long distance charges, because calls spend only a limited amount of time as well as distance on the PSTN.
The reduction in communications cost usually varies from 40% to 70% below what legacy Telcos offer. It is the main benefit of using Interconnect Telecom SIP Trunking. The other major benefits it provides are those which enhance functionality. Some of the features of the Interconnect Telecom SIP Trunking business VoIP product line include discounted international calling, PSTN origination/termination, one BTN (Billing Telephone Number) with E911, unlimited concurrent call sessions (limited only by available bandwidth), operation with legacy CPE through VoIP gateways to deliver IP Communications benefits, additional DIDs to be ordered individually or in any amount, G.711 and G.729A codecs, unlimited local and long distance calling and a 24/7 Network Operations Centre with expert engineering support.
Additional options are also available, such as Calling Name (CNAM), Directory Listing (DL) predicated upon geographical relevance, discounted inbound toll-free (with RespOrg portability), Dynamic load balancing across multiple locations, enhanced 911, business continuity and Local Number Portability (LNP).
Interconnect Telecom's SIP Trunking solutions offer additional savings by eliminating the cost and maintenance associated with troublesome and expensive legacy leased line phone services while reducing long-distance telephone charges. In addition, Interconnect Telecom VoIP Solution SIP Trunking allows VoIP business enterprises to retain their legacy PBX or customer premises equipment (CPE) and connect to SIP Trunking in an evolutionary - rather than revolutionary - fashion!
Session Initiation Protocol (SIP) describes the basic signals used to initiate, manage and terminate communications. SIP Trunking is a term applied to the services offered by LECs (Local Exchange Carriers), ILECs (Independent Local Exchange Carriers), CLECs (Competitive Local Exchange Carriers) and ITSPs (Internet Telephony Service Providers) to terminate Voice over IP (VoIP) calls to the Public Switched Telephone Network (PSTN).
SIP Trunking describes the use of a new or existing high speed internet connection as a replacement for traditional telephone circuits for routing incoming and outgoing calls. Local telephone numbers can be called by people anywhere in the world and rung through to business enterprises using public internet and SIP Trunks. SIP Trunks are, in fact, not Trunks - dedicated circuits - at all. Instead, the number of SIP trunks that a business requires is determined by the number of simultaneous calls rather than a fixed set of wired connections. SIP Trunking goes beyond voice. It is the industry-backed control standard for multimedia communications or convergence. This includes a broad range of communications: voice, data, video and mobile. Because SIP trunks are not fixed in capacity, they can grow rapidly to respond to the needs of your busy enterprise. In addition, SIP Trunks can be delivered anywhere, served by the sufficient levels of Internet connectivity.
How can SIP Trunking be delivered?
Over the Public Internet
It allows any enterprise, anywhere, to adopt SIP Trunking and assign bandwidth to voice at no extra charge for the connection while providing the highest ROI.
The Carrier supplies a connection from their Point of Presence to the enterprise site. This service offers quality of service guarantees. This is a more expensive solution.
In this case, the carrier (LEC, ILEC or CLEC) delivers a managed service using Multi-Protocol Label Switching to ensure the highest voice quality and reliability. The voice quality is very high, even over an unmanaged public Internet connection. Typical savings over PRIs vary from 40-60% with the payback period for the equipment required, which may include an upgrade to the IP-PBX and the installation of an Ingate SIParator or Firewall.