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Merge branch 'feature/add_voip_example' into 'master'
Add voip example See merge request adf/esp-adf-internal!310
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Submodule esp-adf-libs
updated
4 files
+4 −2 | component.mk | |
+205 −0 | esp_sip/include/esp_sip.h | |
+79 −0 | esp_sip/include/g711.h | |
+ − | esp_sip/lib/libesp_sip.a |
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# | ||
# This is a project Makefile. It is assumed the directory this Makefile resides in is a | ||
# project subdirectory. | ||
# | ||
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PROJECT_NAME := voip_app | ||
#CURRENT_PATH := $(shell pwd) | ||
include $(ADF_PATH)/project.mk | ||
#IDF_PATH := $(CURRENT_PATH)/esp-idf | ||
# EXTRA_COMPONENT_DIRS := $(CURRENT_PATH)/components | ||
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#include $(IDF_PATH)/make/project.mk |
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# VoIP Example | ||
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This example allows users to make calls over the internet. | ||
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## Compatibility | ||
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| ESP32-LyraT | ESP32-LyraT-MSC | | ||
|:-----------:|:---------------:| | ||
| [![alt text](../../docs/_static/esp32-lyrat-v4.2-side-small.jpg "ESP32-LyraT")](https://docs.espressif.com/projects/esp-adf/en/latest/get-started/get-started-esp32-lyrat.html) | [![alt text](../../docs/_static/esp32-lyratd-msc-v2.2-small.jpg "ESP32-LyraTD-MSC")](https://docs.espressif.com/projects/esp-adf/en/latest/get-started/get-started-esp32-lyratd-msc.html) | | ||
| ![alt text](../../docs/_static/yes-button.png "Compatible") | ![alt text](../../docs/_static/yes-button.png "Compatible") | | ||
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## Usage | ||
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Prepare the audio board: | ||
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- Connect speakers or headphones to the board. | ||
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Configure the example: | ||
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- Select compatible audio board in `menuconfig` > `Audio HAL`. | ||
- 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`. | ||
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Configure external application: | ||
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Setup the PBX Server like Yet Another Telephony Engine (FreePBX/FreeSwitch or any other PBXs) | ||
http://docs.yate.ro/wiki/Beginners_in_Yate | ||
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## Features | ||
- 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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## Reference | ||
http://www.yate.ro/ | ||
https://www.tutorialspoint.com/session_initiation_protocol/index.htm | ||
https://tools.ietf.org/html/rfc3261 |
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menu "VOIP App Configuration" | ||
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config WIFI_SSID | ||
string "WiFi SSID" | ||
default "myssid" | ||
help | ||
SSID (network name) for the example to connect to. | ||
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config WIFI_PASSWORD | ||
string "WiFi Password" | ||
default "mypassword" | ||
help | ||
WiFi password (WPA or WPA2) for the example to use. | ||
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choice VOIP_CODEC | ||
prompt "SIP Codec" | ||
default G711A | ||
help | ||
sip codec | ||
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config SIP_CODEC_G711A | ||
bool "G711 PCMA" | ||
config SIP_CODEC_G711U | ||
bool "G711 PCMU" | ||
endchoice | ||
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config SIP_URI | ||
string "SIP URI" | ||
default "udp://100:100@192.168.1.123:5060" | ||
help | ||
The SIP URI with username, password, address, port and transport type | ||
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endmenu |
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# | ||
# "main" pseudo-component makefile. | ||
# | ||
# (Uses default behaviour of compiling all source files in directory, adding 'include' to include path.) | ||
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