Why You Need SIP Protocol for Voice Communications
SIP (Session Initiation Protocol) is the communications technology for person-to-person voice traffic over the Web. 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 expanded to deal with IP telephony in conjunction with video and instant messaging. Forthcoming improvements will enable SIP applications like video conference calls, application sharing, home monitoring, and interactive gaming for companies in Phillips.
Producers of SIP hardware are rapidly creating innovative hardware and software to take advantage of this new Internet communications technique. SIP telephones, PC client software, SIP servers, routers, and firewalls are now available from companies such as Ingate Systems and Cisco.
Before installing a SIP plan in your organization, you should consider the proper configuration of your corporate firewall to support SIP. Many of the typical firewalls already deployed in business offices are not developed to support the SIP protocol. First, SIP media streams are transmitted over dynamically assigned UDP ports that are typically shut on firewalls. Second, SIP clients within a firewall can not be reached using IP addresses since 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 correctly support SIP to get around these technical problems. 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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