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Showing posts from September, 2016

Opus negotiation for the practical man

Opus [0] is a versatile audio codec, with a variable sample rate and bitrate, suitable for both music and speech. It is defined in RFC 6716 [1] and required by WebRTC [2]. Opus can operate at various sample rates, from 8 KHz to 48 KHz, and at variable bitrates, from 6 kbit/sec to 510 kbit/sec. The RTP payload format defined for Opus in RFC 7587 [3] explains the use of media type parameters in SDP, and this article aims to analyze them and show in particular how "asymmetric streams" can be achieved. This is an example of SDP defining an Opus offer or answer:       ...

Deploying Homer with Puppet

Fan of Homer ? So am I, and as sometimes happens I'm a fan who could join the team! If despite the title of this post you're still reading, then it's a good sign and we can move on. Homer is a vast project that aims to provide a tool, with a GUI, to correlate all the signalling, RTCP stats, events, logs in your RTC network. It focuses heavily on SIP , for historical reasons, but it's also an extendible framework to store other types of signalling, correlate data, and compute statistics. People browsing their github account are often heard saying "Do they have this too? And this ? Wow!". It is compatible off the shelf with common applications like Kamailio, opensips, FreeSWITCH, Asterisk, so if you're into VoIP, adding Homer to your platform is as easy as installing it and telling your apps where to send their data. There are also standalone tools like captagent ,  nodejs apps to parse and collect specific logs, to be associated with the re...