Inside the Asterisk

Keyword: RTP

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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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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AstriDevCon 2018 – A Recap

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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RTP Security Vulnerabilities: A Retrospective

[dropshadowbox align=”none” effect=”lifted-both” width=”auto” height=”” background_color=”#ffffff” border_width=”1″ border_color=”#dddddd” ]tl;dr: We fixed the vulnerabilities. If you’d like to read the conclusion of this admittedly long saga,

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WebRTC and Asterisk 14

WebRTC has been in Asterisk since Asterisk 11 and over time has evolved just as the WebRTC specification itself has evolved. The WebRTC implementation we

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rtcp-mux in WebRTC

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?

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