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https://github.com/OLIMEX/sip_phone_example
SIP Phone based on Olimex ESP32-ADF, MOD-LCD2.8RTP boards.
https://github.com/OLIMEX/sip_phone_example
Last synced: about 2 months ago
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SIP Phone based on Olimex ESP32-ADF, MOD-LCD2.8RTP boards.
- Host: GitHub
- URL: https://github.com/OLIMEX/sip_phone_example
- Owner: OLIMEX
- Created: 2019-09-10T08:51:57.000Z (over 5 years ago)
- Default Branch: master
- Last Pushed: 2021-03-22T07:33:42.000Z (almost 4 years ago)
- Last Synced: 2024-08-02T19:38:48.818Z (5 months ago)
- Language: C
- Homepage:
- Size: 1.11 MB
- Stars: 45
- Watchers: 8
- Forks: 20
- Open Issues: 3
-
Metadata Files:
- Readme: README.md
Awesome Lists containing this project
README
# 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
# Get ESP-ADF
Install requared packages:
```bash
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:
```bash
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:
```bash
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:
```bash
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