How to implement a WebRTC call?

Question

VoIP

I am writing a bot for one online WebRTC calling service
Data is sent from the server to connect to TURN servers.
Tell me, how can I implement the transmission of audio and receive it from the caller?
You need a ready-made solution or code example in PHP or Node JS.
Packet transmission over SRTP with ZRTP protection.
Ready-made audio files in a suitable format for dubbing by a bot are already on the server.
An example of the config of ICE servers for connection:

Spoiler

"ice_servers": [  
   {  
      "urls":[  
         "stun:59.213.22.11:8443"
      ]
   },
   {  
      "urls":[  
         "turn:59.213.22.11:8443?transport=udp"
      ],
      "username":"1234567891:3d6077641c0ff83f2baf8721657dce03-987654321",
      "credentialType":"password",
      "credential":"Po\/9QZBYu4hsWXCV30Plu+8l9y4="
   },
   {  
      "urls":[  
         "turn:59.213.22.11:8443?transport=tcp"
      ],
      "username":"1234567891:3d6077641c0ff83f2baf8721657dce03-987654321",
      "credentialType":"password",
      "credential":"Po\/9QZBYu4hsWXCV30Plu+8l9y4="
   }
]

Other possibly relevant information:

Spoiler

"zrtp-hash" : "1.10 397ca930186c896dde33b896b793100364151820887f088020a3ad0d4c4dcda1",
"candidate": [
    {
      "generation": 0,
      "ip": "222.111.22.11",
      "name": "audio_rtp",
      "network_name": "wlan0",
      "password": "f13o7dtcakjh316",
      "port": "45591",
      "priority": 1.0,
      "proto": "udp",
      "type": "local",
      "username": "P3O+VUco5nipV1RY"
    }
  ],
"fast_connect": 2,
"jb_flags": 3,
"peerList": [],
"timeoutSec": 60,
"useragent": {
    "caps": 3847,
    "state": 7,
    "ua_ver": "com.adroid.app 1.1",
    "voip_ver": "voip android release version:1.2.0.569 date:Jul 12 2018 19:23:12"
}

Answer 1

freeswitch+sipml5

Answer 2

A bit different from what you asked, but you can try https://voximplant.com