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SIP Phone based on Olimex ESP32-ADF, MOD-LCD2.8RTP boards.
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README.md

SIP Phone Example

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

Compatibility

ESP32-ADF https://www.olimex.com/Products/IoT/ESP32/ESP32-ADF/open-source-hardware

ESP32-ADF

Get ESP-ADF

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 https://github.com/espressif/esp-adf.git
cd esp-adf
git submodule update --init
export ADF_PATH=$PWD
cd esp-idf
./install.sh
. ./export.sh

Usage

Prepare the audio board:

  • Connect speakers or headphones to the board.

Load the example:

cd ~/
mkdir ~/espwork
cd ~/espwork
git clone --recursive https://github.com/OLIMEX/sip_phone_example.git
cd  sip_phone_example
git submodule update --init
cp lvgl_component.mk components/lvgl/component.mk
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) http://docs.yate.ro/wiki/Beginners_in_Yate

Features

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

Reference

http://www.yate.ro/ https://www.tutorialspoint.com/session_initiation_protocol/index.htm https://tools.ietf.org/html/rfc3261

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