Why You Need SIP Protocol for Voice Communications
Session Initiation Protocol (SIP) is the transmission protocol for person-to-person voice data 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 handle IP telephony combined with video and instant messaging. Future enhancements will enable SIP applications like video conference calls, application sharing, home monitoring, and interactive gaming for businesses in Hermann.
Suppliers of SIP equipment are rapidly creating innovative hardware and software to get the upper hand in this new Internet communications technique. SIP telephones, PC client software, SIP servers, routers, and firewalls are now available from companies such as Ingate Systems and Cisco.
Prior to setting up a SIP solution in your business, you should consider the proper settings of your corporate firewall to accept SIP. Many of the typical firewalls already installed in business offices are not developed to allow the SIP protocol. First, SIP media streams are transferred over dynamically allocated UDP ports that are usually blocked 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 provider or Ethernet fiber service is able to support the SIP protocol from your internal network to the outside world. Your IT manager will need to ascertain how to correctly support SIP to get around these technical problems. By including a SIP proxy and registrar for managing the firewall, it is feasible to handle complex SIP scenarios for reliable and confidential communications.
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