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SIP Phone based on Olimex ESP32-ADF, MOD-LCD2.8RTP boards.
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components sync to lattest esp-adf Mar 11, 2020
main sync to lattest esp-adf Mar 11, 2020
.gitignore Create .gitignore Sep 10, 2019
.gitmodules Add lvgl submodule Sep 10, 2019
Makefile initial upload Sep 10, 2019 Update Mar 12, 2020 initial upload Sep 10, 2019
partitions_voip_example.csv initial upload Sep 10, 2019
sdkconfig sync to lattest esp-adf Mar 11, 2020
sdkconfig.defaults initial upload Sep 10, 2019

SIP Phone Example

This example allows users to make calls over the internet. The project is basic, but you are free to evolute.





Install requared packages:

sudo apt-get install git wget flex bison gperf python python-pip python-setuptools python-serial python-click python-cryptography python-future python-pyparsing python-pyelftools cmake ninja-build ccache libffi-dev libssl-dev

Download and install esp-adf:

cd ~/
git clone --recursive
cd esp-adf
git submodule update --init
export ADF_PATH=$PWD
cd esp-idf
. ./


Prepare the audio board:

  • Connect speakers or headphones to the board.

Load the example:

cd ~/
mkdir ~/espwork
cd ~/espwork
git clone --recursive
cd  sip_phone_example
git submodule update --init
cp components/lvgl/
make menuconfig

Configure the example:

  • Select compatible audio board in menuconfig > Audio board select.
  • Set up Wi-Fi connection by running menuconfig > VOIP App Configuration and filling in WiFi SSID and WiFi Password.
  • Select compatible audio codec in menuconfig > VOIP App Configuration > SIP Codec.
  • Create the SIP extension, ex: 100 (see below)
  • Set up SIP URI in menuconfig > VOIP App Configuration > SIP_URI.

Upload the example:

make flash monitor

Configure external application:

Setup the PBX Server like Yet Another Telephony Engine (FreePBX/FreeSwitch or any other PBXs)


  • Lightweight
  • Support multiple transports for SIP (UDP, TCP, TLS)
  • Support G711A/8000 & G711U/8000 Audio Codec
  • Easy setting up by using URI


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