Why You Need SIP Protocol for Voice Communications
SIP (Session Initiation Protocol) is the transport technology for person-to-person voice 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 handle IP telephony in conjunction with video and instant messaging. Future enhancements will allow SIP applications like video conference calls, application sharing, home monitoring, and interactive gaming for businesses in Lantana.
Makers of SIP hardware are rapidly creating innovative products and software to get the upper hand in this new Internet communications technique. 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 business, you should think about the proper configuration of your corporate firewall to accept SIP. Many of the typical firewalls already deployed in business offices are not designed to allow the SIP protocol. First, SIP media streams are transmitted over dynamically assigned 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 unique to the LAN. Third, you need to ensure that either your T1 line carrier or Metro 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 correctly support SIP to get around these technical issues. By including a SIP proxy and registrar for controlling the firewall, it is feasible to handle complicated SIP scenarios for reliable and confidential communications.
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