By Flavio E. Goncalves
It is a functional, hands-on publication established round a fictitious case examine VoIP supplier that you'll construct on a improvement server utilizing OpenSIPS 1.6. The case research grows bankruptcy through bankruptcy, from fitting your neighborhood improvement server, correct as much as the completed VoIP supplier. This booklet is for readers who are looking to know how to construct a SIP supplier from scratch utilizing OpenSIPS. it really is appropriate for VoIP services, huge companies, and universities. Telephony and Linux adventure could be priceless yet isn't really crucial. Readers don't need to have past wisdom of OpenSIPS. This e-book also will support readers who have been utilizing OpenSER yet at the moment are stressed with the hot OpenSIPS.
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Additional info for Building Telephony Systems with OpenSIPS 1.6
These components can reside on the same machine or be spread over several machines, depending upon your dimensioning. One of these components is the SIP proxy server; in our case, the server running the OpenSIPS software. What best describes the OpenSIPS server is a SIP router. It is capable of manipulating the SIP headers and routing packets at extremely high speeds. Third-party modules give OpenSIPS extreme flexibility to play roles that were not originally intended, such as NAT traversal, IMS, load balancing, and other functionalities.
Normally, the INVITE message contains the SDP offer message, while "200 Ok" contains the answer message. These messages are shown in the next screenshot. You can observe that the GSM codec is offered, but the other phone does not support it. 729. The session rtpmap:101 is the DTMF relay described in the RFC2833. [ 22 ] This material is copyright and is licensed for the sole use by Betty Vaughan-Pope on 1st February 2010 2601 S Broadway St, Unit 29, La Porte, , 77571 Chapter 1 Here is the INVITE (SDP offer) message: The following screenshot shows the "200 Ok" (SDP answer) reply: [ 23 ] This material is copyright and is licensed for the sole use by Betty Vaughan-Pope on 1st February 2010 2601 S Broadway St, Unit 29, La Porte, , 77571 Introduction to SIP The SIP protocol and the OSI model It is always important to understand the voice protocols against the OSI model to situate where each protocol fits.
Now, there are two variants—Kamailio and OpenSIPS. I don't want to get into the politics of SER, Kamailio, and OpenSIPS. The concepts presented here are valid for both. I wrote the book using OpenSIPS because, in my opinion, they are delivering software in a consistent manner. Another decision is to write about the development version. I hope it will be released by the time the book is available. OpenSIPS has a flexible plugin model for third-party applications. Applications can be easily created and plugged into the server.
Building Telephony Systems with OpenSIPS 1.6 by Flavio E. Goncalves