Why You Need SIP Protocol for Voice Communications
SIP (Session Initiation Protocol) is the transmission technology for person-to-person voice data over the Internet. 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 expanded to handle IP telephony in conjunction with video and instant messaging. Future enhancements will allow SIP applications such as video conference calls, application sharing, home monitoring, and interactive gaming for businesses in Gowanda.
Makers of SIP hardware are quickly creating innovative hardware 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.
Prior to implementing a SIP plan in your organization, you should think about the appropriate configuration of your corporate firewall to accept SIP. Many of the typical firewalls currently installed in business offices are not designed to allow the SIP protocol. First, SIP media streams are transferred over dynamically allocated UDP ports that are often closed on firewalls. Second, SIP clients within a firewall can not be accessed using IP addresses since these addresses are local and unique to the LAN. Third, you need to ensure that either your T1 line broadband carrier or Metro fiber provider is able to support the SIP protocol from your internal network to the outside world. Your IT administrator will need to ascertain how to properly support SIP to overcome these technical difficulties. By adding a SIP proxy and registrar for controlling the firewall, it is feasible to deal with complicated SIP scenarios for reliable and private communications.
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