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rtcp-mux in WebRTC

By Mark Michelson

Do you use WebRTC with Asterisk? Did you notice calls stop working after updating Google Chrome to version 57? Are you curious why that happened? The answer is the rtcp-mux [...]

Asterisk, Opus, packet loss, and FEC

By Kevin Harwell

So you’ve heard there is now an Opus codec for Asterisk that’s been released. However, you are having problems with poor audio quality due to packets being dropped or lost. [...]

Dialplan handler routines allow customization

By Richard Mudgett

There are several handler routines available to allow you to customize behavior for the different states of a call. Handler routines execute outside of the normal dialplan execution flow. It [...]

Five Shocking Tips on How to Get Asterisk Bugs Resolved Quickly

By Rusty Newton

Warning, these tips may not be shocking, but you already know that and you clicked anyway! Shame! Many of us know the frustrating feeling of submitting a bug report and seeing it [...]

Pjproject 2.6 now qualified for use with Asterisk!

By George Joseph

This week, we’re pleased to say that we’ve updated the Asterisk 13, 14 and master branches’ bundled version of pjproject to 2.6. [...]

You Down With SDP? (Yeah you know me)

By Mark Michelson

In the previous post, Josh introduced the forthcoming addition of streams to Asterisk. I’m going to piggyback on that to introduce a unified SDP API to Asterisk. What’s the motivation [...]

Stream Support in Asterisk

By Joshua Colp

Media streams are something that we all use every day. From watching videos on Youtube to placing calls media streams are right there with us in the background. They are [...]

Configuring the Opus Encoder for Asterisk

By Kevin Harwell

The recently announced Opus codec for Asterisk exposes a few configuration options that allow you to manipulate the encoder for your particular setup. These options can be set within codecs.conf. [...]

Jitter Buffer Operation and Use in Asterisk

By Kevin Harwell

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 [...]

SIP timers T1 and B affect performance

By Richard Mudgett

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 [...]

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