Consider SIP Protocol for Voice Communications
Session Initiation Protocol (SIP) is the transport 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 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 combined with video and instant messaging. Future improvements will enable SIP applications such as video conference calls, application sharing, home monitoring, and interactive gaming for companies in Osage Beach.
Producers of SIP components are quickly creating innovative hardware 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 available from companies such as Ingate Systems and Cisco.
Prior to setting up a SIP solution in your company, you should take into account the proper settings of your corporate firewall to accept 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 assigned UDP ports that are normally shut on firewalls. Second, SIP clients within a firewall can not be accessed 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 administrator will need to ascertain how to correctly support SIP to overcome these technical issues. By adding a SIP proxy and registrar for controlling the firewall, it is possible to deal with complicated SIP scenarios for secure and confidential communications.
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