Consider SIP Protocol for Voice Communications
Session Initiation Protocol (SIP) is the transport technology 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 handle IP telephony combined with video and instant messaging. Forthcoming enhancements will allow SIP applications such as video conference calls, application sharing, home monitoring, and interactive gaming for companies in Stanislaus.
Manufacturers of SIP components are quickly crafting new hardware and software to take advantage of 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.
Before setting up a SIP solution in your organization, you should think about the appropriate settings of your corporate firewall to accept SIP. Many of the typical firewalls already deployed in business offices are not developed to support the SIP protocol. First, SIP media streams are transferred over dynamically allocated UDP ports that are normally blocked on firewalls. Second, SIP clients within a firewall can not be reached using IP addresses because these addresses are local and unique to the LAN. Third, you need to ensure that either your T1 line carrier or Ethernet 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 properly support SIP to overcome these technical challenges. By including a SIP proxy and registrar for controlling the firewall, it is feasible to handle complicated SIP scenarios for reliable and confidential communications.
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