Connect call from Asterisk to Google STT using PHPAGI

johnbj · May 30, 2020, 5:42pm

Hi am using PHPAGI I want to use the Google API below to convert live audio from Asterisk and return text.

Like if a user calls and gets to Asterisk, I want the call to be connected to the Google API below to do live recording and convert into text. Is this possible to connect a call from Asterisk to the API below using PHPAGI and I any guide on how to implement this?

If this is not possible what are other option like using JAVA with Google STT and AGI?

github.com

GoogleCloudPlatform/php-docs-samples/blob/master/speech/src/streaming_recognize.php

<?php
/**
 * Copyright 2016 Google Inc.
 *
 * 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.
 */

/**
 * For instructions on how to run the full sample:
 *

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ambiorixg12 · May 31, 2020, 5:08am

Totally possible I done many times, just focus to get google API working, and then just pass the recorded audio to speech recognition script,

chrcaes · May 31, 2020, 9:25am

Hi Johnbj,

Im using Zaf script and it works perfect. Also developing my own api. Feel free to contact me if u have any doubts:

github.com

zaf/asterisk-speech-recog/blob/master/speech-recog.agi

#!/usr/bin/env perl

#
# AGI script that renders speech to text using Google's Cloud Speech API.
#
# Copyright (C) 2011 - 2016, Lefteris Zafiris <zaf@fastmail.com>
#
# This program is free software, distributed under the terms of
# the GNU General Public License Version 2. See the COPYING file
# at the top of the source tree.
#
# -----
# Usage
# -----
# agi(speech-recog.agi,[lang],[timeout],[intkey],[NOBEEP],[rtimeout],[speechContexts])
# Records from the current channel until 2 seconds of silence are detected
# (this can be set by the user by the 'timeout' argument, -1 for no timeout) or the
# interrupt key (# by default) is pressed. If NOBEEP is set, no beep sound is played
# back to the user to indicate the start of the recording. If 'rtimeout' is set, 
# overwrite to the absolute recording timeout. 'SpeechContext' provides hints to

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nshmyrev · May 31, 2020, 10:29am

Hi, this is very suboptimal solution since your users will experience delays first for VAD to detect silence, then for Google to process speech. The delays could be several seconds. The zaf script also suffers from this design flow. And, moreover you have to pay for the Google.

The right solution would be to stream audio with EAGI/ARI/Asterisk speech module to python server probably and then you’ll get the results immediately.

You can check the following EAGI python code using Vosk server for example:

github.com

alphacep/vosk-server/blob/master/client-samples/asterisk/eagi.py

#!/usr/bin/python3

from asterisk.agi import *
import os
from websocket import create_connection
import json
import traceback

AUDIO_FD = 3
CONTENT_TYPE = 'audio/l16; rate=8000; channels=1'
ACCEPT = 'audio/pcm'

def process_chunk(agi, ws, buf):
    agi.verbose("Processing chunk")
    ws.send_binary(buf)
    res = json.loads(ws.recv())
    agi.verbose("Result: " + str(res))
    if 'result' in res:
        text = " ".join([w['word'] for w in res['result']])
        os.system("espeak -w /tmp/response22.wav \"" + text.encode('utf-8') + "\"")

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This code works with offline Vosk server and doesn’t require keys or internet connection, but it is also possible to set code with Google streaming API, I am just not aware of any examples of such setup.

system · June 30, 2020, 10:30am

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Connect call from Asterisk to Google STT using PHPAGI

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