Why You Need SIP Protocol for Voice Over IP
Session Initiation Protocol (SIP) is the transport protocol for person-to-person voice traffic over the Internet. 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 Williams.
Producers of SIP hardware are rapidly developing innovative hardware and software to get the upper hand in this new Internet communications technique. SIP telephones, PC client applications, SIP servers, routers, and firewalls are now obtainable from companies such as Ingate Systems and Cisco.
Before implementing a SIP solution in your company, you should take into account the appropriate configuration of your corporate firewall to accept SIP. Many of the common firewalls already 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 often closed on firewalls. Second, SIP clients within a firewall can not be reached 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 issues. By adding a SIP proxy and registrar for managing the firewall, it is possible to handle complicated SIP scenarios for secure and confidential communications.
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