Why You Need SIP Protocol for Voice Over IP
Session Initiation Protocol (SIP) is the transmission 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 expanded to handle IP telephony combined with video and instant messaging. Forthcoming improvements will allow SIP applications such as video conference calls, application sharing, home monitoring, and interactive gaming for companies in Whatcom.
Makers of SIP components are rapidly creating innovative products and software to get the upper hand in this new Internet communications method. SIP telephones, PC client programs, SIP servers, routers, and firewalls are now on the market from companies such as Ingate Systems and Cisco.
Prior to setting up a SIP solution in your organization, you should take into account the proper settings of your corporate firewall to accept SIP. Many of the typical 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 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 Ethernet fiber carrier is able to handle the SIP protocol from your internal network to the outside world. Your IT manager will need to evaluate how to appropriately support SIP to overcome these technical problems. By adding a SIP proxy and registrar for controlling the firewall, it is possible to deal with complicated SIP situations for secure and private communications.
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