Inside the Asterisk

RTP: retransmission for video to combat packet loss

Introduction Packet loss can be an annoying problem when dealing with real time communication, especially when dealing with video. It’s very noticeable when the screen freezes for multiple seconds, then the footage resumes with everything in a completely different position than it was originally. We’ve all seen this before. Packet loss is inevitable, but it

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It’s getting to be that time of year!

With the changing weather, some of us are thinking about summer with excitement, others about winter with trepidation.  Regardless of which hemisphere you live in though, it’s time to start thinking about Asterisk 16!

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A Call to Action!

Hey Everybody! For those of you that don’t know me, my name is Matthew Fredrickson and I’m the Asterisk Open Source Project Lead. Does that mean I’m the best software developer or most prolific committer to the project? No. Does it mean I can answer every single question about using Asterisk? Not necessarily. Oftentimes it

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WebRTC and Asterisk: When It Goes Wrong

Like many things WebRTC is a complex stack of technology within Asterisk and also within the browser. The result of this is that to the best of our ability it doesn’t always work. The browser can change things, the network can stop things from working, the Javascript client may have an issue. This blog post

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Asterisk Call Party, Privacy, and Header Presentation

Asterisk allows users to manipulate call party identification information through mechanisms like configuration options and dialplan functions (for instance CALLERID and CONNECTEDLINE to name a couple). This grants the user freedom to adjust values with regards to what call/caller information to expose and/or override. With this freedom, though, comes some complexity, and confusion. Especially when

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The Woes of TLS Certificates and WebRTC

TLS certificates and their management are something we take for granted every day when we visit a website. If you sit down and try to explain to someone how it all fits together however it is quite easy to overwhelm them. It is complicated to understand how it all fits together but it’s the foundation

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Identifying an endpoint in PJSIP

A basic concept with chan_pjsip/res_pjsip is the endpoint. When a new SIP request comes in, res_pjsip needs to identify which endpoint the request is for. New incoming SIP requests are identified by various endpoint identifiers registered with res_pjsip. The various endpoint identifiers look for different things in the received request to determine which endpoint is

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WANTED: Dead or Alive!

The Story of Asterisk and Keep-Alives The vast majority of VoIP communications is done via UDP datagrams.  It’s a no-overhead protocol which makes it fast and although it also makes it unreliable, the SIP and RTP protocols and our own ears and eyes can tolerate a certain amount of packet loss quite easily.   From a

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PJSIP: DNS Manager (dnsmgr) and Full Dynamic Hostname Support, Coming Soon!

Recently there’s been discussion on chan_sip going away in the future which led to many comparisons between it and chan_pjsip. What does chan_pjsip do better? What is chan_pjsip missing? What does chan_sip do better? What does chan_pjsip have that chan_sip doesn’t? Enter dnsmgr for PJSIP One thing chan_sip has that chan_pjsip was lacking is the

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CDR/CEL Processing – Climbing the Beanstalk

One of the most annoying tasks within Asterisk (or VoIP in general) is the task of CDR and event processing. Why is processing these so annoying? Well, depending on your infrastructure, problems can arise because of any of the below: Row locking within the database Handling of multiple input points Handling a constantly changing data

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