Why You Need SIP Protocol for Voice Over IP
Session Initiation Protocol (SIP) is the transmission technology for person-to-person real time traffic 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 such as video conference calls, application sharing, home monitoring, and interactive gaming for companies in Union.
Manufacturers of SIP hardware are rapidly designing new hardware and software to take advantage of 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.
Before implementing a SIP plan in your company, you should consider the appropriate configuration of your corporate firewall to support SIP. Many of the common firewalls already installed in business offices are not designed to allow the SIP protocol. First, SIP media streams are transmitted over dynamically allocated UDP ports that are often shut on firewalls. Second, SIP clients inside a firewall can not be accessed using IP addresses since these addresses are local and unique to the LAN. Third, you need to ensure that either your T1 line carrier or Metro fiber carrier is able to handle the SIP protocol from your internal network to the outside world. Your IT administrator will need to evaluate how to appropriately support SIP to get around these technical problems. By adding a SIP proxy and registrar for controlling the firewall, it is possible to handle complicated SIP situations for reliable and private communications.
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