By Flavio E. Goncalves
It is a functional, hands-on ebook established round a fictitious case research VoIP supplier that you're going to construct on a improvement server utilizing OpenSIPS 1.6. The case learn grows bankruptcy by way of bankruptcy, from fitting your neighborhood improvement server, correct as much as the completed VoIP supplier. This e-book is for readers who are looking to know the way to construct a SIP supplier from scratch utilizing OpenSIPS. it's compatible for VoIP prone, huge organizations, and universities. Telephony and Linux adventure can be necessary yet isn't really crucial. Readers needn't have previous wisdom of OpenSIPS. This e-book also will support readers who have been utilizing OpenSER yet at the moment are burdened with the recent OpenSIPS.
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Extra resources for Building Telephony Systems with OpenSIPS 1.6
Branch routing blocks: Contains the logic to be executed for each branch of the SIP request, just before forwarding it out. Local routing blocks: Local routing blocks are executed when OpenSIPS internally generates a request (acting as UAS only) using the Transaction Module (TM) . Error routing block: This route is executed when a parsing error for a SIP request is detected. cfg file will be covered in detail in the chapters ahead. cfg file is a script which is executed for each SIP message received.
Determine the request's target. 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)?
In this example, userA uses an IP phone to call another IP phone over the network. To complete the call, two SIP proxies are used. userA calls userB using its SIP identity, called the SIP URI. com. com. A call made using sips: (secure SIP) will use a secure transport (TLS-Transport Layer Security) between the caller and the callee. The transaction starts with userA sending an INVITE request addressed to userB. The INVITE request contains a certain number of header fields. Header fields are named attributes that provide additional information about the message.
Building Telephony Systems with OpenSIPS 1.6 by Flavio E. Goncalves