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 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 extended to handle IP telephony in conjunction with video and instant messaging. Future enhancements will enable SIP applications such as video conference calls, application sharing, home monitoring, and interactive gaming for companies in Houston.
Producers of SIP equipment are quickly creating new products and software to take advantage of this new Internet communications technique. SIP telephones, PC client software, SIP servers, routers, and firewalls are now obtainable from companies such as Ingate Systems and Cisco.
Prior to setting up a SIP plan in your business, 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 support the SIP protocol. First, SIP media streams are transmitted over dynamically assigned UDP ports that are usually shut on firewalls. Second, SIP clients within a firewall can not be reached using IP addresses since these addresses are local and private to the LAN. Third, you need to ensure that either your T1 line broadband carrier or Ethernet fiber provider is able to support the SIP protocol from your internal network to the outside world. Your IT administrator will need to ascertain how to properly support SIP to get around these technical issues. By adding a SIP proxy and registrar for controlling the firewall, it is possible to handle complicated SIP scenarios for reliable and confidential communications.
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