Developing for Kamailio locally using Docker.

 The plan was to participate in KWC2020 in the flesh. But pandemic add a correction to this plan. 

Talk on how to have different development environements on a local PC/Mac using Docker and docker-compose.

 

 And links from the presentation:

1. https://github.com/igorolhovskiy/kamailio-dev-dc - Look for the branches

2. https://docs.docker.com/config/containers/runmetrics/

3. https://engineering.squarespace.com/blog/2017/understanding-linux-container-scheduling

4. https://sysdig.com/blog/container-isolation-gone-wrong/

5. https://discuss.linuxcontainers.org/t/cpu-limits-confusion/1837

6. https://docs.docker.com/config/containers/resource_constraints/#configure-the-realtime-scheduler

7. https://lwn.net/Articles/549580/

FusionPBX - VTiger 7 Integration

Every telecom engineer should build a PBX, CallCenter, and CRM Integration.

So, my time for CRM integration came. The idea is quite simple.

1. Calls are registering in CRM with call recording
2. Calls are being notified with popup while using CRM
3. Calls from CRM directly. Click 2 Call actually

Integration is done for popular opensource CRM VTiger 7 and FusionPBX. Have to say, not plain, reworked, my fork actually.

This project consists of 3 parts.

1. PBXManager  - module for VTiger CRM, built on top of the original PBXManager module made for Asterisk.
2. VFusion Daemon - Docker Image that is running with FusionPBX (FreeSwitch) and listening to FreeSwitch events and making requests to VTiger CRM. Made on NodeJS
3. FusionPBX - Actually my fork of it. Version 4.4

I actually tried to avoid daemons or so, resides on FusionPBX server, but really can't avoid it totally. So, architecture is not perfect. Some of the requests are made from FreeSwitch via mod_curl, some - in v_xml_cdr app by Fusion, some - in VFusion daemon. Not clean, but as of now it's working.
So, how it looks like?

1. Call from unknown contact

2. Call from a known contact

 

3. Call details records

yes, you can listen to the recordings directly from CRM.


Click 2 Call is made quite simple. Just click on a phone number inside CRM and you will have a callback to your extension. After picking up, a call to that number would be established.

As it's intended to be a commercial product, code is open. But documentation is still not. Maybe in the future documentation also would be open.

If you have any interest in deploying this product - pls contact me at support <at> consertis.at. Or here.

Janus Audio/Video plays around

WebRTC got to me. I can't say it was the first time, but most deep at the moment.

Small disclaimer. I'm not a web developer and only now got, that browser is OS by itself.

And the way to get media from the browser is WebRTC. So, let's make one more bicycle to have WebRTC - VoIP bridge.

There are many projects around to fill this gap. FreeSwitch (mode_verto or mod_sofia), Kamailio + RTPEngine are just to mention some.

And Janus was chosen. To try something new and get some skills in bright'n'shiny world of web development. And I like, that Janus has already built-in SIP plugin.

The idea of this small proof of concept - get audio and video streams separated and treated separately. Why - to have audio part handled by SIP (passed to Asterisk, more precisely) and video (or screen sharing part in the future) - by Janus SFU unit (VideoRoom plugin)

Project is just a set of 3 docker containers running in network host mode (just don't want to play around with NAT and all this stuff) and glued together with docker-compose. Nginx as a web server, Janus as Janus WebRTC server, and Asterisk as SIP application server.

Web part is just reworked and gloriously copy-pasted from original Janus demos.

This project is intended to be used as a playground for future experiments and donor of code for other projects.

Repo with project on GitHub.

Some of the pictures not to be so boring:

Audio is passed to Asterisk via SIP plugin, and Video is handled by EchoTest plugin.

Audio is mixing by Asterisk (SIP plugin as well), Video is handled by VideoRoom plugin.

Point, this one is tested on a VPS with a Let'sEncrypt certs to test it mostly "in the wild" and also there are many restrictions in modern browsers about WebRTC and security, which means even over LAN you need to have pure HTTPS.

Building RTPEngine with Docker

Idea is quite simple.
To build rtpengine packet on Debian system, but not having all build dependencies on target machine.
So, we'll use Docker to generate deb files for target system and than - just remove image. This way system will held only minimum needed packets.

As a target, Debian 10 (buster) will be used, as I succeed to install rtpengine mr8.0 (with G.729 support) on it and can't get it working on Debian 9 (maybe use lower version?)

So. All is done on Debian 10 netinst.

• First step - install Docker
  
  apt update
  apt install -y apt-transport-https ca-certificates curl gnupg2 software-properties-common
  curl -fsSL https://download.docker.com/linux/debian/gpg | apt-key add -
  add-apt-repository "deb [arch=amd64] https://download.docker.com/linux/debian $(lsb_release -cs) stable"
  apt update
  apt install -y docker-ce
  
• Prepare Dockerfile
  
  FROM debian:buster
  MAINTAINER Igor Olhovskiy <igorolhovskiy@gmail.com>
  
  ENV DEBIAN_FRONTEND noninteractive
  ENV RTPENGINE_VERSION mr8.0
  ENV BCG729_VERSION 1.0.4
  
  RUN apt-get update && \
      apt-get upgrade -y && \
      apt-get install -y git \
                  dpkg-dev \
                  cmake \
                  unzip \
                  wget \
                  debhelper-compat \
                  default-libmysqlclient-dev \
                  gperf \
                  iptables-dev \
                  libavcodec-dev \
                  libavfilter-dev \
                  libavformat-dev \
                  libavutil-dev \
                  libbencode-perl \
                  libcrypt-openssl-rsa-perl \
                  libcrypt-rijndael-perl \
                  libcurl4-openssl-dev \
                  libdigest-crc-perl \
                  libdigest-hmac-perl \
                  libevent-dev libglib2.0-dev \
                  libhiredis-dev libio-multiplex-perl \
                  libio-socket-inet6-perl libiptc-dev \
                  libjson-glib-dev libnet-interface-perl \
                  libpcap0.8-dev \
                  libpcre3-dev \
                  libsocket6-perl \
                  libspandsp-dev \
                  libssl-dev \
                  libswresample-dev \
                  libsystemd-dev \
                  libxmlrpc-core-c3-dev \
                  markdown \
                  curl \
                  wget \
                  zlib1g-dev && \
      cd /usr/src && \
      curl https://codeload.github.com/BelledonneCommunications/bcg729/tar.gz/$BCG729_VERSION > bcg729_$BCG729_VERSION.orig.tar.gz && \
      tar zxf bcg729_$BCG729_VERSION.orig.tar.gz && \
      cd bcg729-$BCG729_VERSION && \
      git clone https://github.com/ossobv/bcg729-deb.git debian && \
      dpkg-buildpackage -us -uc -sa && \
      cd /usr/src && \
      dpkg -i *.deb && \
      cd /usr/src && \
      git clone -b $RTPENGINE_VERSION https://github.com/sipwise/rtpengine.git && \
      cd rtpengine && \
      dpkg-buildpackage && \
      apt-get clean && \
      rm -rf /var/lib/apt/lists/*
  
  RUN mkdir -p /opt/deb && \
      mv /usr/src/*.deb /opt/deb
  
  VOLUME ["/opt/deb"]
• Build and get deb files
  
  docker build -t rtpengine:build .
  docker create --name rtpengine-build rtpengine:build
  docker cp rtpengine-build:/opt/deb .
• Install rtpengine
  
  
  cd deb
  dpkg -i libbcg729-0*.deb
  dpkg -i ngcp-rtpengine-daemon_*.deb
  apt-get install -f
  dpkg -i ngcp-rtpengine-recording-daemon_*.deb
  apt-get install -f
  dpkg -i ngcp-rtpengine-utils_*.deb
  apt-get install -f
  dpkg -i ngcp-rtpengine-iptables_*.deb
  apt-get install -f
  dpkg -i ngcp-rtpengine-kernel-dkms_*.deb
  dpkg -i ngcp-rtpengine_*.deb
  Idea of repeating apt-get install -f is to apt to add missing packets.
  Next - configure rtpengine as usual