Why You Need SIP Protocol for Voice Communications
SIP (Session Initiation Protocol) is the transmission protocol for person-to-person voice data 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 combined with video and instant messaging. Forthcoming enhancements will enable SIP applications such as video conference calls, application sharing, home monitoring, and interactive gaming for businesses in Airville.
Suppliers of SIP hardware are rapidly creating new hardware and software to take advantage of this new Internet communications method. SIP telephones, PC client programs, SIP servers, routers, and firewalls are now on the market from companies such as Ingate Systems and Cisco.
Before setting up a SIP solution in your company, you should take into account the appropriate settings of your corporate firewall to support SIP. Many of the typical firewalls currently 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 normally shut on firewalls. Second, SIP clients within 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 broadband provider or Metro fiber service is able to support the SIP protocol from your internal network to the outside world. Your IT manager will need to evaluate how to appropriately support SIP to overcome these technical issues. By adding a SIP proxy and registrar for controlling the firewall, it is possible to deal with complicated SIP situations for secure and confidential communications.
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