This project is made to provide a simple Audio an Video WebRTC to SIP gateway using the WebRTC possibility of new Astersik versions. It's based on https://github.com/asterisk/cyber_mega_phone_2k
This gateway was made to easily connect a browser to any classic SIP endpoint like Softphone, PABX or MCU.
https://github.com/asterisk/cyber_mega_phone_2k https://wiki.asterisk.org/wiki/display/AST/Installing+Asterisk+From+Source https://wiki.asterisk.org/wiki/display/AST/WebRTC+tutorial+using+SIPML5 https://webrtc.ventures/2018/03/webrtc-sip-the-demo/
sh install.shDamien Fétis
Licensed under the Apache License, Version 2.0 (the "License"); you may not use this file except in compliance with the License. You may obtain a copy of the License at
http://www.apache.org/licenses/LICENSE-2.0
Unless required by applicable law or agreed to in writing, software distributed under the License is distributed on an "AS IS" BASIS, WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. See the License for the specific language governing permissions and limitations under the License.