Why You Need SIP Protocol for Voice Communications
SIP (Session Initiation Protocol) is the communications technology for person-to-person real time 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 deal with IP telephony combined with video and instant messaging. Future enhancements will allow SIP applications like video conference calls, application sharing, home monitoring, and interactive gaming for companies in Delhi.
Producers of SIP hardware are rapidly crafting 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 on the market from companies such as Ingate Systems and Cisco.
Before implementing a SIP solution in your organization, you should think about the proper configuration of your corporate firewall to accept SIP. Many of the typical firewalls currently 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 often closed on firewalls. Second, SIP clients inside a firewall can not be accessed using IP addresses because these addresses are local and private to the LAN. Third, you need to ensure that either your T1 line provider 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 adding a SIP proxy and registrar for managing the firewall, it is feasible to handle complicated SIP situations for secure and private communications.
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