Why You Need SIP Protocol for Voice Communications
SIP (Session Initiation Protocol) is the transmission 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 extended to handle IP telephony in conjunction with video and instant messaging. Future enhancements will allow SIP applications such as video conference calls, application sharing, home monitoring, and interactive gaming for companies in Kossuth.
Makers of SIP equipment are rapidly developing new hardware and software to take advantage of this new Internet communications method. SIP telephones, PC client applications, SIP servers, routers, and firewalls are now on the market from companies such as Ingate Systems and Cisco.
Before implementing a SIP solution in your organization, you should take into account the proper configuration of your corporate firewall to support SIP. Many of the common firewalls already installed in business offices are not designed to support the SIP protocol. First, SIP media streams are transmitted over dynamically assigned UDP ports that are usually blocked on firewalls. Second, SIP clients within a firewall can not be reached 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 Metro fiber carrier is able to support the SIP protocol from your internal network to the outside world. Your IT administrator will need to evaluate how to properly support SIP to overcome these technical issues. By including a SIP proxy and registrar for controlling the firewall, it is feasible to deal with complex SIP situations for secure and confidential communications.
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