Why You Need SIP Protocol for Voice Over IP
SIP (Session Initiation Protocol) is the communications protocol for person-to-person real time traffic over the Internet. Its defining 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 deal with 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 Lyon.
Suppliers of SIP hardware are rapidly developing new hardware and software to take advantage of 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.
Before implementing a SIP solution in your organization, you should take into account the proper configuration 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 transmitted over dynamically allocated UDP ports that are usually blocked on firewalls. Second, SIP clients inside 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 carrier 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 evaluate how to correctly support SIP to get around these technical difficulties. By including a SIP proxy and registrar for managing the firewall, it is possible to deal with complicated SIP scenarios for secure and private communications.
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