Why You Need SIP Protocol for Voice Communications
Session Initiation Protocol (SIP) is the communications technology for person-to-person real time data over the Internet. Its defining specifications come from the SIP working group of the Internet Engineering Task Force. SIP gives 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 extended to handle IP telephony in conjunction with video and instant messaging. Forthcoming improvements will allow SIP applications such as video conference calls, application sharing, home monitoring, and interactive gaming for companies in Pyatt.
Makers of SIP components are rapidly designing new products and software to get the upper hand in this new Internet communications technique. SIP telephones, PC client software, SIP servers, routers, and firewalls are now on the market from companies such as Ingate Systems and Cisco.
Prior to installing a SIP plan in your organization, you should think about the appropriate configuration of your corporate firewall to support SIP. Many of the typical firewalls currently installed in business offices are not developed to allow the SIP protocol. First, SIP media streams are transmitted over dynamically allocated 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 broadband provider or Metro fiber service is able to support the SIP protocol from your internal network to the outside world. Your IT administrator will need to ascertain how to properly support SIP to get around these technical difficulties. By including a SIP proxy and registrar for controlling the firewall, it is feasible to deal with complex SIP scenarios for reliable and confidential communications.
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