By Flavio E. Goncalves
It is a sensible, hands-on ebook dependent round a fictitious case examine VoIP supplier that you're going to construct on a improvement server utilizing OpenSIPS 1.6. The case learn grows bankruptcy by means of bankruptcy, from fitting your neighborhood improvement server, correct as much as the completed VoIP supplier. This ebook is for readers who are looking to know how to construct a SIP supplier from scratch utilizing OpenSIPS. it really is compatible for VoIP prone, huge organisations, and universities. Telephony and Linux adventure should be priceless yet isn't crucial. Readers needn't have past wisdom of OpenSIPS. This publication also will support readers who have been utilizing OpenSER yet at the moment are pressured with the hot OpenSIPS.
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Additional info for Building Telephony Systems with OpenSIPS 1.6
SIP proxy The SIP proxy is the central component of our solution. It is responsible for registering the users and keeping the location database (maps IP to SIP addresses). The entire SIP routing and signaling is handled by the SIP proxy, and it is also responsible for end-user services such as call forwarding, white/blacklist, speed dialing, and others. This component never handles the media (RTP packets); all media-related packets are routed directly from the user agent clients, servers, and PSTN gateways.
4. ProxyB consults their own location database for userB's address, and then adds another VIA header field with their own address to the INVITE request and forwards it to userB's IP address. 5. userB's phone receives the INVITE request and starts ringing. The phone indicates back this condition by sending a message of "180 Ringing". 6. This message is routed back through both proxies in the reverse direction. Each proxy uses the Via header field to determine where to send the response and removes their own Via header from the top.
4. Forward the request to the target. 5. Process all responses. A stateful proxy creates a new server transaction for each new request received. Any retransmissions of the request will then be handled by that server transaction. Example: For each request traversing our SIP proxy, we will: Step 1: Request validation • Check the message size to avoid buffer overflows • Check the Max-forwards header to detect loops Step 2: Routing information preprocessing • If a record-route header exists, process it Step 3: Determine the request target • Is it in the location database (registered users)?