Why You Need SIP Protocol for Voice Over IP
SIP (Session Initiation Protocol) is the transmission protocol for person-to-person real time traffic over the Internet. 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 allow SIP applications like video conference calls, application sharing, home monitoring, and interactive gaming for businesses in Boise.
Producers of SIP hardware are rapidly developing innovative hardware and software to take advantage of this new Internet communications scheme. SIP telephones, PC client programs, SIP servers, routers, and firewalls are now available from companies such as Ingate Systems and Cisco.
Before installing a SIP plan in your organization, you should take into account the proper settings of your corporate firewall to support SIP. Many of the common firewalls already installed in business offices are not developed to allow the SIP protocol. First, SIP media streams are transmitted over dynamically assigned UDP ports that are usually shut on firewalls. Second, SIP clients within 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 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 evaluate how to appropriately support SIP to overcome these technical difficulties. By adding a SIP proxy and registrar for managing the firewall, it is possible to deal with complex SIP situations for secure and confidential communications.
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