Consider SIP Protocol for Voice Communications
SIP (Session Initiation Protocol) is the transport technology for person-to-person real time 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 extended to handle IP telephony in conjunction with video and instant messaging. Future improvements will allow SIP applications such as video conference calls, application sharing, home monitoring, and interactive gaming for companies in Orange.
Suppliers of SIP equipment are quickly crafting new 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 implementing a SIP plan in your company, you should consider the appropriate settings of your corporate firewall to accept SIP. Many of the common firewalls currently installed in business offices are not designed to support the SIP protocol. First, SIP media streams are transmitted over dynamically allocated UDP ports that are normally closed 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 Ethernet fiber carrier is able to handle the SIP protocol from your internal network to the outside world. Your IT administrator will need to ascertain how to correctly support SIP to get around these technical problems. By adding a SIP proxy and registrar for controlling the firewall, it is feasible to deal with complex SIP situations for reliable and confidential communications.
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