Inside the Asterisk

Keyword: sip

PJSIP Transport Reload Changes

In a past blog post I talked about how you can’t reload transports without enabling explicit support, but alluded to changes coming in the future.

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A Video Update

Video has been a continued theme of Asterisk for some years now. We put into place the foundation to allow us to do video better,

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STIR/SHAKEN in Asterisk

What is STIR/SHAKEN? STIR/SHAKEN (Secure Telephony Identity Revisited / Signature-based Handling of Asserted information using toKENs) is a new technology that the telecommunications industry is

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Common SIP URI Issues

When transitioning from the chan_sip channel driver to chan_pjsip one of the items that can catch people off guard is the use of SIP URIs

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SIP and RTP Routing

One of the most common issues I see when people deploy SIP is calls hanging up after approximately 30 seconds or traffic not going to

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An Asterisk Video Update

Over the past few years we’ve been working to improve the video support in Asterisk. We initially started with adding stream support[1] in a backwards

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PJSIP Configuration Design

A comment that I see frequently when helping people with PJSIP is the lack of a general section (with global options) and how this causes

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