Why You Need SIP Protocol for Voice Communications
Session Initiation Protocol (SIP) is the communications technology for person-to-person voice data 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 deal with IP telephony combined with video and instant messaging. Forthcoming enhancements will enable SIP applications like video conference calls, application sharing, home monitoring, and interactive gaming for companies in Cloverdale.
Producers of SIP hardware are rapidly creating innovative products and software to get the upper hand in this new Internet communications method. 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 settings of your corporate firewall to accept SIP. Many of the common firewalls currently deployed in business offices are not developed to allow the SIP protocol. First, SIP media streams are transmitted over dynamically allocated UDP ports that are typically blocked 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 Ethernet fiber service is able to support the SIP protocol from your internal network to the outside world. Your IT administrator will need to evaluate 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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