Why You Need SIP Protocol for Voice Communications
Session Initiation Protocol (SIP) is the communications protocol for person-to-person real time data over the Web. Its technical specifications come from the SIP working group of the Internet Engineering Task Force. SIP provides access to the public switched telephone network for voice data at 3 kHz bandwidth and common number dialing using Voice over IP (VoIP). It can also be expanded to handle IP telephony in conjunction with video and instant messaging. Forthcoming enhancements will allow SIP applications like video conference calls, application sharing, home monitoring, and interactive gaming for businesses in Mcpherson.
Suppliers of SIP components are quickly developing innovative products and software to get the upper hand in this new Internet communications method. SIP telephones, PC client applications, SIP servers, routers, and firewalls are now on the market from companies such as Ingate Systems and Cisco.
Before installing a SIP plan in your organization, you should take into account the appropriate settings of your corporate firewall to accept SIP. Many of the typical firewalls already deployed in business offices are not developed to support the SIP protocol. First, SIP media streams are transferred over dynamically assigned UDP ports that are usually closed on firewalls. Second, SIP clients within a firewall can not be reached using IP addresses because these addresses are local and private to the LAN. Third, you need to ensure that either your T1 line provider or Metro fiber carrier is able to support the SIP protocol from your internal network to the outside world. Your IT administrator will need to evaluate how to properly support SIP to get around these technical difficulties. By adding a SIP proxy and registrar for controlling the firewall, it is possible to handle complicated SIP scenarios for secure and private communications.
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