Why You Need SIP Protocol for Voice Over IP
SIP (Session Initiation Protocol) is the transmission protocol for person-to-person voice traffic over the Web. 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 deal with IP telephony combined with video and instant messaging. Forthcoming improvements will enable SIP applications such as video conference calls, application sharing, home monitoring, and interactive gaming for businesses in Chambers.
Producers of SIP components are rapidly designing innovative products and software to take advantage of this new Internet communications method. SIP telephones, PC client programs, SIP servers, routers, and firewalls are now obtainable from companies such as Ingate Systems and Cisco.
Prior to implementing a SIP plan in your company, you should consider the appropriate settings of your corporate firewall to support SIP. Many of the common firewalls already installed in business offices are not developed to allow the SIP protocol. First, SIP media streams are transmitted over dynamically allocated UDP ports that are normally shut on firewalls. Second, SIP clients within 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 broadband provider or Metro fiber service 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 managing the firewall, it is possible to deal with complicated SIP scenarios for secure and confidential communications.
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