Why You Need SIP Protocol for Voice Over IP
SIP (Session Initiation Protocol) is the transmission protocol for person-to-person voice traffic over the Web. Its defining 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 combined with video and instant messaging. Future enhancements will enable SIP applications such as video conference calls, application sharing, home monitoring, and interactive gaming for companies in Glen Mills.
Manufacturers of SIP equipment are quickly designing new hardware and software to get the upper hand in this new Internet communications method. 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 setting up a SIP solution in your company, you should think about the appropriate configuration of your corporate firewall to support SIP. Many of the common firewalls currently installed in business offices are not designed to support the SIP protocol. First, SIP media streams are transmitted over dynamically assigned UDP ports that are usually shut on firewalls. Second, SIP clients within a firewall can not be reached using IP addresses since these addresses are local and unique to the LAN. Third, you need to ensure that either your T1 line broadband carrier or Metro fiber carrier is able to support the SIP protocol from your internal network to the outside world. Your IT manager 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 feasible to handle complicated SIP situations for reliable and private communications.
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