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, and have over time taken advantage of this and advanced things further. I thought I would take a little bit of time to reflect back on what has been done.
Inside the Asterisk
Codec negotiation in Asterisk has been one of its deepest darkest secrets. It’s been around since the beginning and over the past two decades it’s grown and mutated into one of the least understood parts of Asterisk. With Advanced Codec Negotiation that’s about to change! One of the Asterisk team’s goals for 2020 was to dig
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 using to help combat telephony fraud. We’ve all received spam calls, and some of us may have even received a call from a caller ID we recognized, but ended up
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 within PJSIP. This is because in chan_sip these are generated on your behalf based on different configuration options while in chan_pjsip we leave this up to the user. Let’s take
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 with the call starting out with video. This is a use case which is useful for a lot of people and has worked well. Coming soon, however, is the ability
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 where it should. This can be hard for users to grasp and is primarily due to the fact that SIP embeds routing information (IP addresses and ports) within the signaling
Over the past few years we’ve been working to improve the video support in Asterisk. We initially started with adding stream support in a backwards compatible fashion so we could individually address streams and add/remove them. Next we added support for REMB to be able to control the video bitrate with supported clients. We continued
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 chan_pjsip, which has been available in Asterisk since version 12 was released in 2013, you’ll never see this message. But, if you do see this message, do not be afraid!
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 their configuration to be larger than it needs to be. I thought I would take this blog post to explain some of the design choices that went into PJSIP configuration
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 today. To that end the implementation was written to use OpenSSL memory buffers. These are places for OpenSSL to place received data or for OpenSSL to consult when sending a