Why You Need SIP Protocol for Voice Communications
Session Initiation Protocol (SIP) is the transport protocol for person-to-person real time traffic over the Internet. Its technical 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 combined with video and instant messaging. Forthcoming improvements will allow SIP applications such as video conference calls, application sharing, home monitoring, and interactive gaming for businesses in Saint Landry.
Manufacturers of SIP components are quickly creating innovative hardware 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 available from companies such as Ingate Systems and Cisco.
Before installing a SIP solution in your company, you should consider the appropriate configuration of your corporate firewall to support SIP. Many of the common firewalls currently deployed in business offices are not developed to allow the SIP protocol. First, SIP media streams are transferred over dynamically assigned UDP ports that are often shut on firewalls. Second, SIP clients within a firewall can not be reached using IP addresses because these addresses are local and unique to the LAN. Third, you need to ensure that either your T1 line broadband carrier 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 evaluate how to correctly support SIP to get around these technical issues. By adding a SIP proxy and registrar for controlling the firewall, it is possible to deal with complex SIP scenarios for reliable and private communications.
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