Why You Need SIP Protocol for Voice Communications
Session Initiation Protocol (SIP) is the transmission protocol for person-to-person voice traffic over the Web. Its defining 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 expanded to deal with IP telephony combined with video and instant messaging. Future enhancements will allow SIP applications like video conference calls, application sharing, home monitoring, and interactive gaming for companies in Ozan.
Producers of SIP components are rapidly creating new 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 obtainable from companies such as Ingate Systems and Cisco.
Before implementing a SIP solution in your business, you should consider the proper settings of your corporate firewall to support SIP. Many of the typical firewalls already installed in business offices are not developed to support the SIP protocol. First, SIP media streams are transmitted over dynamically assigned UDP ports that are normally closed on firewalls. Second, SIP clients within a firewall can not be reached using IP addresses since these addresses are local and private to the LAN. Third, you need to ensure that either your T1 line provider or Metro fiber carrier is able to handle 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 challenges. By adding a SIP proxy and registrar for controlling the firewall, it is feasible to handle complex SIP situations for secure and confidential communications.
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