Inside the Asterisk

Category: PJSIP

PJSIP Transport Reload Fun

When PJSIP support in Asterisk was being developed one of the critical areas of development was transports. These are for the most part provided by PJSIP and are what allow the flow of SIP signaling. PJSIP provides UDP, TCP, and TLS transports and we provide one for Websockets for WebRTC. Naturally we needed to allow

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PJSIP Invite Session Lifetime

In the past month I’ve been fixing an issue with Asterisk and PJSIP that I thought would be fun to share in a blog post. The originally filed issue was for a crash experienced when Asterisk was manipulating the reference count of a PJSIP invite session. For those who may be unaware the INVITE session

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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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Collaborating for Success in Open Source

Open source is becoming very prevalent in the software world, even if it’s not obvious. Your phone, your television, your smart speaker, and even your car is likely to use open source libraries and applications. In fact, a recent Tidelift survey showed that 92% of applications use open source libraries. One thing I’ve seen over

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Dana and Asterisk, part 2

A couple of weeks ago, Dan Jenkins kindly wrote a guest blog post about Dana — an up-and-coming open source project which helps to highlight some of the great video-conferencing capabilities in Asterisk. In this blog post, I’d like to expand on that, and show you how to get a simple video-conferencing solution up and

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Introducing Dana the Stream Gatekeeper

In this troubling time of dealing with COVID-19 around the world we’re seeing more and more need for tools to help in communicating with co-workers, friends and family. Asterisk has historically proven itself as one of the key puzzle pieces when it comes to enacting change and evolution in the communications space. The power of

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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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