Why You Need SIP Protocol for Voice Communications
SIP (Session Initiation Protocol) is the communications technology for person-to-person real time traffic 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 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 businesses in Follett.
Manufacturers of SIP hardware are rapidly designing new products 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.
Prior to setting up a SIP solution in your organization, you should think about the appropriate configuration 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 transferred over dynamically allocated UDP ports that are usually closed on firewalls. Second, SIP clients inside a firewall can not be accessed using IP addresses since these addresses are local and private to the LAN. Third, you need to ensure that either your T1 line carrier or Ethernet 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 properly support SIP to overcome these technical difficulties. By including a SIP proxy and registrar for controlling the firewall, it is feasible to deal with complicated SIP situations for reliable and private communications.
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