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 Web. 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 in conjunction with video and instant messaging. Future improvements will allow SIP applications like video conference calls, application sharing, home monitoring, and interactive gaming for businesses in Orange.
Manufacturers of SIP components are quickly creating new 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 on the market from companies such as Ingate Systems and Cisco.
Prior to setting up a SIP plan in your business, you should take into account the proper settings of your corporate firewall to accept SIP. Many of the typical firewalls already deployed in business offices are not developed to allow the SIP protocol. First, SIP media streams are transferred over dynamically allocated UDP ports that are often 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 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 ascertain how to appropriately support SIP to get around these technical issues. By including a SIP proxy and registrar for controlling the firewall, it is possible to handle complex SIP situations for secure and private communications.
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