Why You Need SIP Protocol for Voice Over IP
SIP (Session Initiation Protocol) is the communications 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 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. Forthcoming improvements will allow SIP applications such as video conference calls, application sharing, home monitoring, and interactive gaming for businesses in Santa Ynez.
Makers of SIP components are rapidly developing innovative products and software to get the upper hand in this new Internet communications method. 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 company, you should take into account the proper settings of your corporate firewall to accept SIP. Many of the common firewalls already deployed in business offices are not designed to support the SIP protocol. First, SIP media streams are transferred over dynamically assigned UDP ports that are normally shut on firewalls. Second, SIP clients within a firewall can not be reached 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 carrier is able to support the SIP protocol from your internal network to the outside world. Your IT manager will need to ascertain how to appropriately support SIP to overcome these technical problems. By including a SIP proxy and registrar for managing the firewall, it is possible to deal with complicated SIP situations for secure and private communications.
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