Why You Need SIP Protocol for Voice Communications
Session Initiation Protocol (SIP) is the transport 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 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 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 Wedowee.
Producers of SIP hardware are rapidly developing innovative hardware and software to take advantage of this new Internet communications method. SIP telephones, PC client applications, SIP servers, routers, and firewalls are now available from companies such as Ingate Systems and Cisco.
Before implementing a SIP plan in your company, you should take into account the proper configuration of your corporate firewall to support SIP. Many of the typical firewalls already deployed in business offices are not designed to support the SIP protocol. First, SIP media streams are transferred over dynamically allocated UDP ports that are typically shut on firewalls. Second, SIP clients inside 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 provider or Metro fiber provider 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 get around these technical issues. By including a SIP proxy and registrar for managing the firewall, it is feasible to deal with complex SIP scenarios for reliable and private communications.
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