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 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 deal with IP telephony in conjunction with video and instant messaging. Future improvements will enable SIP applications like video conference calls, application sharing, home monitoring, and interactive gaming for businesses in Trumbull.
Suppliers of SIP components are quickly developing new hardware and software to take advantage of this new Internet communications scheme. SIP telephones, PC client software, SIP servers, routers, and firewalls are now on the market from companies such as Ingate Systems and Cisco.
Before installing a SIP plan in your organization, you should think about the proper settings of your corporate firewall to accept SIP. Many of the common firewalls currently installed in business offices are not designed to allow the SIP protocol. First, SIP media streams are transferred over dynamically assigned UDP ports that are often closed on firewalls. Second, SIP clients within a firewall can not be reached using IP addresses because these addresses are local and private to the LAN. Third, you need to ensure that either your T1 line carrier or Metro fiber provider is able to handle 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 problems. By adding a SIP proxy and registrar for controlling the firewall, it is feasible to deal with complex SIP scenarios for reliable and confidential communications.
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