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Energyefficient WebRTC Asterisk systems

Mapping the energy difference between a mesh and a SFU video mode setup

Setup and Installation

One can setup astrisk from theor online manual setup instriction [5] or use the dockerfile given along with this project to setup a SFU webrtc enabled conferencing sip and media server.

Configuration

httpd

  [general]
  enabled=yes
  bindaddr=0.0.0.0
  bindport=8088               ; Port to bind to for HTTP sessions (default is 8088)
  tlsenable=yes               ; enable tls - default no.
  tlsbindaddr=0.0.0.0:8089    ; address and port to bind to - default is bindaddr and port 8089.
  tlscertfile=/etc/asterisk/keys/asterisk.crt
  tlsprivatekey=/etc/asterisk/keys/asterisk.key

Check status on console

*CLI> http show status HTTP Server Status: Prefix: Server: Asterisk Server Enabled and Bound to 0.0.0.0:8088

HTTPS Server Enabled and Bound to 0.0.0.0:8089

Enabled URI's: /httpstatus => Asterisk HTTP General Status /phoneprov/... => Asterisk HTTP Phone Provisioning Tool /metrics/... => Prometheus Metrics URI /ari/... => Asterisk RESTful API /ws => Asterisk HTTP WebSocket

Enabled Redirects: None.

Check status on web

mage

Endpoints with webrtc mage

dialplan in /etc/asterisk/extensions.conf

  exten => 7000,1,Answer()
  same => n,ConfBridge(sfuconfbridge)
  same => n,Hangup()

Description of the endpoints in pjssip

  [sfuconfbridge]
  type=aor
  max_contacts=5
  remove_existing=yes
  
  [sfuconfbridge]
  type=auth
  auth_type=userpass
  username=altanai
  password=password
  
  [sfuconfbridge]
  type=endpoint
  aors=sfuconfbridge
  auth=sfuconfbridge
  dtls_auto_generate_cert=yes
  webrtc=yes
  ; Setting webrtc=yes is a shortcut for setting the following options:
  ; use_avpf=yes
  ; media_encryption=dtls
  ; dtls_verify=fingerprint
  ; dtls_setup=actpass
  ; ice_support=yes
  ; media_use_received_transport=yes
  ; rtcp_mux=yes
  context=default
  direct_media=no
  allow=!all,ulaw,vp8,h264
  max_audio_streams = 10
  max_video_streams = 10

Note To enable multi-stream support in the PSJIP channel set max_audio_streams and max_video_streams options for a given endpoint to any number between 2 and 16 ( highest)

Description of conf applications in /etc/asterisk/confbridge.conf

  [default_bridge]
  type=bridge
  video_mode=sfu

Note : other types of Video modes besides SFU include first_marked and last_marked. Where the first and last used user to join the conference with video capabilities is the single source of video distribution among all participant Also includes 'none' for no source and 'follow_talker' which very interestingly looks for audio activity before switching the video to active speaker.

For direct media

direct_media is used to enable p2p transport in extensions.conf

  [altanai]
  type=endpoint
  aors=altanai
  auth=altanai
  dtls_auto_generate_cert=yes
  webrtc=yes
  context=default
  direct_media=yes
  allow=!all,ulaw,vp8,h264
  max_audio_streams = 10
  max_video_streams = 10

And pjssip.conf

  direct_media=yes       ; Determines whether media may flow
  ; directly between endpoints (default: "yes")

Tools Used for the work

  1. Webrtc client : For the Webrtc SIP user agent to communicate with the SIP and media Server ( asterisk) I configured cyber_mega_phone_2k [1].

  2. Compute Pressure API : For the experiments to record the CPU load and energy efficiency of the RTP topologies namely meshed, mixed and single forwarding I have used the Compute Pressure API [2] To activate goto chrome://flags/ and enable #enable-experimental-web-platform-features

    mage

  3. Docker image for asterisk setup on docker engine [3]

    mage

  4. Asterisk with PJSIP-pjproject

  5. tcpdump to monitor the active calls

     tcpdump -s 0 udp port 5060 -w /home/ubuntu/sipserver_1.pcap

References