Getting a live_transcript_of_your_call_using_the_ari
1. GETTING A LIVE TRANSCRIPT
OF YOUR CALL
PASCAL CADOTTE MICHAUD | CORE DEVELOPER
ASTRICON, ATLANTA, OCT. 2019
2. WHO AM I?
OPEN SOURCE ENTHUSIAST
PASCAL CADOTTE MICHAUD, CORE DEVELOPER @ WAZO
10+ YEARS IN DEVELOPMENT
PROGRAMMING SOFTWARES USING ASTERISK FOR ALMOST 10 YEARS USING
● AMI
● DIALPLAN
● AGI
● ARI AND STASIS APPLICATIONS
@PCADOTTE @PC-M
3. WAZO PLATFORM
PROJECT MISSION
LEVERAGE THE BEST OPEN SOURCE COMPONENTS TO
TRANSFORM THE WAY WE BUILD IP TELECOM
INFRASTRUCTURE AND CREATE COMMUNICATION SERVICES
4. WHY WOULD YOU WANT TO STREAM THE AUDIO OUT OF ASTERISK
5. INTERESTING USES CASES
WHY WOULD I STREAM SOUNDS OUT OF ASTERISK?
● ADDING SUBTITLES TO A CALL
● TRANSLATIONS
● DETECTING CUSTOMER FEELING
● CALL PRIORITIZATION
● FRAUD DETECTION
● INDEXATION
● CALL ROUTING IVR
http://www.wazo-platform.org/blog/solving-the-emergency-call-prioritization-issue-with-programmable-telecom
6. TIMELINE
AUGUST 2018
AUDIO SOCKET
OCTOBER 2018
ASTRIDEVCON
JANUARY 2019
WAZO PARIS HACKATHON
RES_ARI_STREAM
JULY 2019
ASTRICON CFP
OCTOBER 2019
ASTERISK 16.6.0 WITH
EXTERNALMEDIA
OUR JOURNEY
7. AUDIO SOCKET
BY SEÁN C. MCCORD
“AUDIOSOCKET IS A SIMPLE TCP-BASED PROTOCOL FOR SENDING AND
RECEIVING REALTIME AUDIO STREAMS.
THERE EXISTS A PROTOCOL DEFINITION, A GO LIBRARY, AND AN ASTERISK
APPLICATION.”
VERY SIMPLE PROTOCOL USING AN ASTERISK DIALPLAN APPLICATION.
SEE SEÁN’S PRESENTATION TOMORROW AT 10:30 FOR MORE INTERESTING
DETAILS.
https://wiki.asterisk.org/wiki/display/AST/AudioSocket
8. A FIRST WORKING IMPLEMENTATION
RES_ARI_STREAM
RES_ARI_STREAM IS A MODULE
DEVELOPED AS A HACKATHON
PROJECT TO GET A GOOD FEEL OF THE
PROBLEM.
IT USES THE ARI WEBSOCKET WITH
THE SUB-PROTOCOL
“STREAM-CHANNEL”.
9. HOW DOES IT WORK
RES_ARI_STREAM
1. ON LOAD ADDS A SUB-PROTOCOL TO THE WEBSOCKET
2. ON SUBSCRIPTION A HOOK ON RECEIVED FRAMES IS
ADDED
3. ON VOICE FRAME THE VOICE IS TRANSCODED TO SLIN16
4. THEN THE FRAMES ARE WRITTEN TO THE SUBSCRIBED
WEBSOCKET
https://github.com/sboily/wazo-hackathon-asterisk-stream-module
10. A VERY BASIC EXAMPLE
WRITING THE AUDIO FRAMES TO A FILE
import websocket
import sys
with open("out.wav", "wb") as f:
ws = websocket.WebSocketApp(
"ws://localhost:5039/ws",
on_message=lambda ws, msg: out.write(msg),
on_error=lambda ws, error: print(error),
on_close=lambda ws: print(“### closed ###”),
subprotocols=["stream-channel"],
header=[“Channel-ID: “ + sys.argv[1]],
)
ws.run_forever()
11. A VERY SIMPLE EXAMPLE
USAGE
RECORDING THE AUDIO TO A FILE
LISTENING TO YOUR RECORDING
play -t raw -b 16 --endian little -c 1 -e signed-integer -r 16000 out.wav
python3 record-channel-id.py “<my-channel>”
12. ASTERISK IMPLEMENTATION IN ASTERISK 16.6.0
POST /channels/externalmedia
1. CREATE AN EXTERNAL MEDIA CHANNEL WITH AN HOST AND PORT
2. ASTERISK WILL SEND RTP TO THE HOST AND PORT
3. ADD THE EXTERNAL MEDIA CHANNEL TO THE BRIDGE YOU WISH TO LISTEN
4. READ THE RTP FROM THE SOCKET
THE UNICASTRTP_LOCAL_ADDRESS AND UNICASTRTP_LOCAL_PORT CHANNEL
VARIABLES CONTAIN WHAT YOU NEED TO SEND MEDIA BACK TO ASTERISK.
https://wiki.asterisk.org/wiki/display/AST/External+Media+and+ARI
https://github.com/asterisk/asterisk-external-media
13. DIFFERENCES BETWEEN THE 3 OPTIONS
RES_ARI_STREAM ARI EXTERNALMEDIA AUDIOSOCKET
MEDIA DIRECTION UNI-DIRECTIONAL BI-DIRECTIONAL BI-DIRECTIONAL
USAGE
HTTP ENDPOINT WITH A
SPECIFIC SUB-PROTOCOL FOR
THE WEBSOCKET
LIMITED TO CHANNELS IN
STASIS
DIALPLAN APPLICATION AND
CHANNEL DRIVER
ARI RESOURCE IN A GERRIT PULL
REQUEST
FORMAT SLIN16 IS RECEIVED
RTP WITH SPECIFIED CODEC
IS RECEIVED
VERY SIMPLE PROTOCOL WITH
SLIN16 PAYLOAD
CONNECTIVITY CONNECTS TO A CHANNEL CONNECTS TO A BRIDGE
CONNECTS TO A CHANNEL OR A
BRIDGE
TRANSPORT WEBSOCKET UDP TCP
15. DEMO OF THE RES_ARI_STREAM MODULE
https://github.com/pc-m/transcript-demo/blob/master/transcript_demo/wazo.py
https://github.com/pc-m/transcript-demo/blob/master/transcript_demo/transcription.py
INCOMING CALL
AN INCOMING CALL IS
ANSWERED BY ALICE
THIS IS A NORMAL CALL, NO
STASIS APPLICATION IS
REQUIRED HERE
TRANSCRIPT SCRIPT
TRANSCRIPTION SCRIPT IS
LAUNCHED ON THE
INCOMING CALL
THE SCRIPT CONNECTS TO THE
ASTERISK WEBSOCKET AND
REGISTERS ITSELF TO THE
INCOMING CALL CHANNEL ID
GOOGLE SPEECH API
TRANSCRIPTION
GOOGLE SPEECH API IS USED
FOR THE TRANSCRIPTION
USER
HTTP SERVER
THE TRANSLATED TEXT IS
SERVED TO THE USER
17. DEMO OF THE ASTERISK ARI VERSION
https://github.com/pc-m/transcript-demo/blob/master/transcript_demo/ari_stasis.py
https://github.com/pc-m/transcript-demo/blob/master/transcript_demo/ari_server.py
https://github.com/pc-m/transcript-demo/blob/master/transcript_demo/transcription.py
STASIS
THE CALL ENTERS
STASIS
THE STASIS APPLICATION
ADDS THE CALL TO ITS
BRIDGE
UDP SERVER
THE SERVER RECEIVES
THE RTP FROM
ASTERISK
THE SERVER CREATES THE
EXTERNAL MEDIA CHANNELS
WHICH GET ADDED TO THE
STASIS BRIDGE AND RECEIVES
THE RTP
GOOGLE SPEECH API
TRANSCRIPTION
GOOGLE SPEECH API IS
USED FOR THE
TRANSCRIPTION
USER
HTTP SERVER
THE TRANSLATED TEXT IS
SERVED TO THE USER
18. WAZO PLATFORM: COLLABORATION
OPEN COLLABORATION & OPEN DEVELOPMENT
OPEN SOURCE PROJECT
https://www.wazo-platform.org/
PR WORKFLOW @ GITHUB
https://github.com/wazo-platform/
MATTERMOST CHAN
https://mm.wazo.community/wazo-platform/
SERVICE CODE UNDER GLPV3+ LICENCE
LIBRARY CODE UNDER MIT LICENCE
19. LET’S CONNECT
BUILD GREAT SOLUTIONS WITH WAZO PLATFORM
BOOTH WAZO W/ THE TEAM :
FRED, SYLVAIN, PASCAL & JÉROME
TO KNOW MORE :
- JEROME’S TALK : WEDNESDAY - 1:30PM
MORE INFO :
- BLOG POST
- OFFICIAL DEMO @GITHUB
