Softswitch VoIP Solution is one of the major components for growth of VoIP business. It is also one the major reasons for getting good ROI to the VoIP providers and unearthing of so many businesses connected to VoIP. I have shared this article that helps you understand the VoIP Signalling APIs.
With the growing popularity of VoIP systems within corporations as well as in the consumer space, there is a growing need for managers and IT personnel to understand and implement VoIP technology. Setting up a VoIP system may appear terribly complex at first but the basics are easy enough to understand. Before talking about VoIP signaling, we should take a look at the basic infrastructure underlying any VoIP system.
One of the basic components of any VoIP system is the network through which the data travels. Another aspect is the actual media that is to be transported. Signaling is the third important part of making a VoIP call. It can be thought of as an addressing system for VoIP similar to TDM switches in the regular PSTN. Before making a call to someone, the caller needs to know certain information:
1. How and where can the calling party can be reached?
2. Is the other party available right now and if not, are there alternative methods to reach them?
3. Does the other end have the capabilities to support video chat or file transfer?
4. Privacy and authentication issues such as is the other person really the person I want to call?
5. Are other obstacles in the way such as firewalls?
Naturally for two endpoints to be able to communicate, the signaling must be done using the same protocol whether it is SIP, XMPP, H.323, Skype or something else entirely. This - among others - is the basic reason why users on different VoIP networks cannot talk to each other (for example Skype to GTalk). SIP is the most popular protocol at present especially for use within enterprises. Many service vendors offer JavaScript APIs that are fully SIP compatible so that developers can incorporate audio or video calling capabilities easily. For example, the popular mobile platform Android provides a SIP API for use within videoconferencing or IM apps.
This is also where WebRTC and SIP come together. WebRTC is a media engine for browsers with a (soon-to-be) standardized JavaScript API. However, for a call to be made between browsers, a signaling protocol is necessary which is not specified by WebRTC itself. On the other hand, SIP is an open standard for signaling and does not interfere with media transport at all. WebRTC also incorporates some signaling components such as NAT traversal mechanisms - ICE, STUN, TURN etc which are common to SIP as well. Using JavaScript APIs and HTML5, the signaling aspects of SIP and the media capabilities of WebRTC can be easily brought together to make browser to browser calls using completely open standards.
Bhagwad is an expert consultant on business phone systems. He also specializes in WebRTC security.
Article Source: http://EzineArticles.com/?expert=Bhagwad_Park
Article Source: http://EzineArticles.com/8511212
Understanding VoIP Signalling APIs
One of the basic components of any VoIP system is the network through which the data travels. Another aspect is the actual media that is to be transported. Signaling is the third important part of making a VoIP call. It can be thought of as an addressing system for VoIP similar to TDM switches in the regular PSTN. Before making a call to someone, the caller needs to know certain information:
1. How and where can the calling party can be reached?
2. Is the other party available right now and if not, are there alternative methods to reach them?
3. Does the other end have the capabilities to support video chat or file transfer?
4. Privacy and authentication issues such as is the other person really the person I want to call?
5. Are other obstacles in the way such as firewalls?
Naturally for two endpoints to be able to communicate, the signaling must be done using the same protocol whether it is SIP, XMPP, H.323, Skype or something else entirely. This - among others - is the basic reason why users on different VoIP networks cannot talk to each other (for example Skype to GTalk). SIP is the most popular protocol at present especially for use within enterprises. Many service vendors offer JavaScript APIs that are fully SIP compatible so that developers can incorporate audio or video calling capabilities easily. For example, the popular mobile platform Android provides a SIP API for use within videoconferencing or IM apps.
This is also where WebRTC and SIP come together. WebRTC is a media engine for browsers with a (soon-to-be) standardized JavaScript API. However, for a call to be made between browsers, a signaling protocol is necessary which is not specified by WebRTC itself. On the other hand, SIP is an open standard for signaling and does not interfere with media transport at all. WebRTC also incorporates some signaling components such as NAT traversal mechanisms - ICE, STUN, TURN etc which are common to SIP as well. Using JavaScript APIs and HTML5, the signaling aspects of SIP and the media capabilities of WebRTC can be easily brought together to make browser to browser calls using completely open standards.
Bhagwad is an expert consultant on business phone systems. He also specializes in WebRTC security.
Article Source: http://EzineArticles.com/?expert=Bhagwad_Park
Article Source: http://EzineArticles.com/8511212
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