Why You Need SIP Protocol for Voice Communications
Session Initiation Protocol (SIP) is the transport protocol for person-to-person real time data 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 extended to handle IP telephony in conjunction with video and instant messaging. Future enhancements will enable SIP applications like video conference calls, application sharing, home monitoring, and interactive gaming for companies in Fort Belknap Agency.
Makers of SIP components are quickly developing innovative hardware and software to get the upper hand in this new Internet communications scheme. SIP telephones, PC client software, SIP servers, routers, and firewalls are now available from companies such as Ingate Systems and Cisco.
Before setting up a SIP plan in your organization, you should consider the proper settings of your corporate firewall to accept 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 usually shut on firewalls. Second, SIP clients inside a firewall can not be reached 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 provider is able to handle the SIP protocol from your internal network to the outside world. Your IT administrator will need to ascertain how to correctly support SIP to overcome these technical difficulties. By adding a SIP proxy and registrar for managing the firewall, it is possible to handle complicated SIP situations for reliable and confidential communications.
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