Why You Need SIP Protocol for Voice Communications
Session Initiation Protocol (SIP) is the transmission technology for person-to-person real time 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 improvements will enable SIP applications such as video conference calls, application sharing, home monitoring, and interactive gaming for businesses in Thornburg.
Producers of SIP components are rapidly developing innovative products and software to take advantage of this new Internet communications method. SIP telephones, PC client programs, SIP servers, routers, and firewalls are now available from companies such as Ingate Systems and Cisco.
Before setting up a SIP solution in your organization, you should consider the proper settings of your corporate firewall to support SIP. Many of the typical firewalls already deployed in business offices are not designed to allow the SIP protocol. First, SIP media streams are transferred over dynamically allocated UDP ports that are typically closed on firewalls. Second, SIP clients inside a firewall can not be accessed using IP addresses because 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 handle the SIP protocol from your internal network to the outside world. Your IT manager will need to ascertain how to correctly support SIP to overcome these technical difficulties. By including 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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