Why You Need SIP Protocol for Voice Over IP
SIP (Session Initiation Protocol) is the communications protocol 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 in conjunction with video and instant messaging. Forthcoming improvements will enable SIP applications like video conference calls, application sharing, home monitoring, and interactive gaming for companies in Waterloo.
Manufacturers of SIP components are quickly crafting innovative hardware and software to take advantage of this new Internet communications scheme. SIP telephones, PC client applications, SIP servers, routers, and firewalls are now obtainable from companies such as Ingate Systems and Cisco.
Before setting up a SIP solution in your company, you should take into account the appropriate configuration of your corporate firewall to support SIP. Many of the typical firewalls currently deployed in business offices are not designed to allow the SIP protocol. First, SIP media streams are transmitted over dynamically assigned UDP ports that are often 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 broadband carrier or Ethernet fiber service is able to support the SIP protocol from your internal network to the outside world. Your IT manager will need to ascertain how to appropriately support SIP to get around these technical difficulties. By adding a SIP proxy and registrar for controlling the firewall, it is feasible to handle complicated SIP situations for reliable and confidential communications.
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