SIP and RTP Routing
Joshua C. Colp
3 Comments
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
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
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
If you download Asterisk 17 and start it up, you might be one of the people that notices the following messages: If you are using
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
When DTLS support in Asterisk was added the information about how to use DTLS support in OpenSSL was not as flushed out as it is
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.
One of the most difficult things in PJSIP is ensuring that the experience is the best it can be for not just people who configure
[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,
In my previous post I talked about what WebRTC support is like in Asterisk 14. Since Asterisk 15 is going to be released soon let’s
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
