Inside the Asterisk

Jitter Buffer Operation and Use in Asterisk

Jitter buffer functionality has been in Asterisk for quite some time now. However, knowing what jitter is in a voice over IP (VoIP) application and when to use a de-jittering buffer to manage it may still be misunderstood by some. What is a jitter buffer In Asterisk, and generally speaking in VoIP, jitter is the

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SIP timers T1 and B affect performance

Busy Asterisk systems can be affected by the SIP timers T1 and B timeout values configured.  Consideration of their values impacts how quickly a transaction can recover from a lost packet and the amount of memory used.  It is in your best interest to make these values as small as possible for your installation. The

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New Feature: PJSIP Dual Stack

A new feature that was initially implemented during a recent visit to SIPit has now been merged into the 13, 14, and master Asterisk branches. It’s called PJSIP dual stack! For those who may be unfamiliar with what dual stack is it is technique of running both IPv4 and IPv6 connectivity on a system. This

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Asterisk 14 Webinar

Slight interlude from your regularly scheduled programming. For any interested, Matthew Fredrickson, manager of the Asterisk project, will be giving a webinar today about Asterisk 14 and what’s new with Asterisk since the 13 release. You can get info about it at: https://bit.ly/2gDkXyn It will be live today at 8AM, 2PM, and 9PM CDT. Hope to

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The Case Of The Misunderstood PJSIP Callback

When the PJSIP work for Asterisk began one of the primary concerns kept in mind was that it be extensible. One of the APIs derived from this concern was session supplements. Session supplements are a way for modules to add themselves in to the handling of SIP messages for sessions (or calls as you may know

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AstriDevCon 2016 – A Recap

For those of you who were unable to attend, AstriDevCon this year at Phoenix continued to impress. There were 39 people signed up, and at least 43 (and probably more – walkins are common) who actually attended. In terms of structure, AstriDevCon is mostly free form, with the number one item of the day being

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Automatically Download Binary Modules: New Feature in Asterisk 13.12 and 14.1

The Digium Phone Module for Asterisk and the g729a, silk, siren7 and siren14 codec binary modules hosted at downloads.digium.com can now be automatically downloaded and installed during the Asterisk install process. If selected in make menuselect under the ‘External’ sections of ‘Resource Modules’ and ‘Codec Translators’ respectively, ‘make install’ will check the downloads.digium.com web site for the

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Documentation? We’ve Got That!

I’ve been a part of the Asterisk project for many years now, over 10 in fact. I’ve seen it evolve and grow and one of those areas which has changed that I’d like to focus on today is documentation. When Asterisk started out documentation was not as complete or as detailed as it is today

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Asterisk Architecture: The Bridging Framework Part I

Overview When the development goals were laid out at AstriDevCon for Asterisk 12 way back in 2012, we had two primary missions: Build a new SIP channel driver to replace the venerable but aging chan_sip  channel driver. Provide consistency in the APIs exposed by Asterisk so that it is easier to build applications on top of

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Asterisk 14: Publishing Extension State

Asterisk 14 now has the ability to publish extension state using PJSIP PUBLISH requests to another entity acting as an event state compositor. For larger installations, the advantage of this ability is to offload from Asterisk the SUBSCRIBE and NOTIFY responsibility for state changes to the other entity. What can be used as an event

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