Why You Need SIP Protocol for Voice Communications
SIP (Session Initiation Protocol) is the transport protocol for person-to-person real time data over the Web. Its defining 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 deal with IP telephony combined with video and instant messaging. Forthcoming enhancements will allow SIP applications like video conference calls, application sharing, home monitoring, and interactive gaming for businesses in Winthrop Harbor.
Manufacturers of SIP hardware are quickly creating innovative products 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.
Prior to implementing a SIP solution in your organization, you should consider the proper settings of your corporate firewall to support SIP. Many of the typical firewalls currently installed in business offices are not developed to allow the SIP protocol. First, SIP media streams are transmitted over dynamically allocated UDP ports that are normally closed on firewalls. Second, SIP clients within a firewall can not be accessed using IP addresses because these addresses are local and private 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 manager will need to ascertain how to properly 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 complicated SIP scenarios for secure and confidential communications.
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