Why You Need SIP Protocol for Voice Communications
Session Initiation Protocol (SIP) is the communications protocol for person-to-person real time data over the Internet. Its technical specifications come from the SIP working group of the Internet Engineering Task Force. SIP gives 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 enhancements will allow SIP applications like video conference calls, application sharing, home monitoring, and interactive gaming for companies in Queen Creek.
Producers of SIP components are rapidly developing innovative hardware and software to take advantage of this new Internet communications technique. SIP telephones, PC client programs, SIP servers, routers, and firewalls are now obtainable from companies such as Ingate Systems and Cisco.
Before implementing a SIP solution in your organization, 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 designed to support the SIP protocol. First, SIP media streams are transferred over dynamically assigned UDP ports that are usually 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 broadband provider 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 evaluate how to appropriately support SIP to overcome these technical difficulties. By adding a SIP proxy and registrar for controlling the firewall, it is possible to deal with complex SIP situations for secure and private communications.
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