Why You Need SIP Protocol for Voice Communications
Session Initiation Protocol (SIP) is the communications technology for person-to-person real time data 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 deal with IP telephony combined with video and instant messaging. Forthcoming improvements will enable SIP applications such as video conference calls, application sharing, home monitoring, and interactive gaming for businesses in Kokomo.
Producers of SIP hardware are rapidly creating innovative products and software to get the upper hand in this new Internet communications method. SIP telephones, PC client software, SIP servers, routers, and firewalls are now available from companies such as Ingate Systems and Cisco.
Prior to setting up a SIP plan in your company, you should think about the appropriate configuration of your corporate firewall to accept SIP. Many of the common firewalls already deployed in business offices are not developed to allow the SIP protocol. First, SIP media streams are transferred over dynamically assigned UDP ports that are typically blocked 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 provider or Ethernet fiber carrier is able to support the SIP protocol from your internal network to the outside world. Your IT manager will need to evaluate how to properly support SIP to get around these technical problems. By adding a SIP proxy and registrar for controlling the firewall, it is possible to deal with complex SIP scenarios for reliable and private communications.
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