Why You Need SIP Protocol for Voice Communications
Session Initiation Protocol (SIP) is the transport 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 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 companies in Butler.
Suppliers of SIP equipment are quickly developing innovative products and software to get the upper hand in this new Internet communications method. SIP telephones, PC client programs, SIP servers, routers, and firewalls are now available from companies such as Ingate Systems and Cisco.
Prior to setting up a SIP plan in your company, you should consider the appropriate configuration of your corporate firewall to accept SIP. Many of the typical firewalls currently deployed in business offices are not developed to support the SIP protocol. First, SIP media streams are transferred over dynamically assigned UDP ports that are often blocked on firewalls. Second, SIP clients inside a firewall can not be accessed using IP addresses since 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 manager will need to evaluate how to properly support SIP to overcome these technical issues. By adding a SIP proxy and registrar for managing the firewall, it is feasible to handle complex SIP scenarios for secure and private communications.
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