Inside the Asterisk

Category: Architecture

Never Cross the Streams

If you are new to audio over websocket and come from the telephony world, you might assume that this works similarly to an RTP stream

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Asterisk and RFC 4733

Up until recently Asterisk only supported RFC 4733 RTP events when using 8KHz codecs like G.711.  However, with this recent change, Asterisk now supports the

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A Bridge Too Far

In a previous blog post we talked about using Asterisk’s uni-cast functionality as a bridge between the PSTN and an external service.  This post entails

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Asterisk Stir-Shaken Refactor

Note: This new implementation is available as of Asterisk 18.22.0, 20.7.0, and 21.2.0. It’s been almost 4 years since STIR/SHAKEN support was first added to

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Speeding up ARI

As you may have seen from a recent submission, it was recently found that there was a consistent delay when adding a channel into the

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