Never Cross the Streams
Mike Bradeen
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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
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
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
A recurring theme I’m seeing lately is people deploying VoIP, running into issues, and not approaching their issues from the perspective of taking all components
Objectively measuring call quality has always been tough because it’s, well, highly subjective. The first measurement process that usually comes to mind is the Mean
If you have ever tried to use direct media with Asterisk, you my have received this message before. It can happen when the User Agent
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,
When stream support was added to Asterisk it was initially done with the focus being for SFU with a single video stream from each participant
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
Hey Everybody, It’s about a month out from AstriDevCon 2018 and I wanted to write a little bit to summarize what we discussed this year.
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