Inside the Asterisk

Keyword: pjsip

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, 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.

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Asterisk ACN: Advanced Codec Negotiation

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

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STIR/SHAKEN in Asterisk

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

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Common SIP URI Issues

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

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New PJSIP Logging Functionality

When PJSIP support was written for Asterisk we naturally needed the ability to display the SIP messages being sent and received. This took the form of the res_pjsip_logger module which hooks into the message sending and receiving path and logs the messages. The functionality was written to be familiar to users of chan_sip by allowing

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Adding and Removing Media Streams

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

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PJSIP Body Generator Persistence

When PJSIP publish and subscribe functionality was created we knew we wanted to provide a pluggable mechanism to allow modules to easily extend and add new bodies. The result of this is what is known as body generators. Given a set of data they convert it into a format expected by a device, such as

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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 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

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AstriDevCon 2019 – A Recap

Greetings All, Time certainly flies and it seems like AstriDevCon and AstriCon 2019 were a century ago. It’s hard to imagine that in fact they were only a month and a half ago! Since everyone has had some time to recharge I thought now was a great time to review and talk about what was

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An Asterisk Video Update

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 compatible fashion so we could individually address streams and add/remove them. Next we added support for REMB[2] to be able to control the video bitrate with supported clients. We continued

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