Never Cross the Streams
Mike Bradeen
May 21, 2025
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
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
One of the historical issues we’ve had with Asterisk and the Asterisk Testsuite has been that if you’ve created an Asterisk pull request for some
Did you know that sipp can play back RTP streams? A lot of long time users may not! Did you know that you can use
If you didn’t already know, both SIP and HTTP share the same digest authentication mechanism described all the way back in RFC-2069 “An Extension to HTTP
Overview When recording audio, it can be useful to split streams from one another. For example, if you are in a call with someone, you
As mentioned in this post, Asterisk now supports the use of RFC4733 digits in common bitrates beyond 8kHz. At the end of the post, we
AstriCon 2025 registration is now open! You can take advantage of early bird pricing by registering here. AstriCon this year will span a total of three
Overview A new feature called Tenant ID (or multi-tenant identifier) was recently added to Asterisk. It doesn’t do anything on its own, but can be
The ability to send SIP NOTIFY messages to endpoints and arbitrary uri’s via AMI and CLI has existed for some time within Asterisk. Changes from