As voice system developers, we often manually test different use cases. Testing with multiple actors can be complicated (i.e., Alice calls Bob, who adds Carol to the call).

Here is a micro-tool based on the PJSUA SIP client.

The name is inspired by an alien from the famous computer game.

1. Build it for the first time:

   docker-compose build --force-rm --no-cache --pull talkingpet

2. Configure the tool by creating an .env file from the provided .env-sample.

3. Execute one of the desired scripts:

   • register SIP_USERNAME: Register a SIP client and allow the tester to control it using keyboard shortcuts.
   • answer SIP_USERNAME: Register a SIP client and automatically answer incoming calls.
   • play SIP_USERNAME /path/to/the/audio.wav: Register a SIP client, answer incoming calls, and play back the given WAV file.
   • dial SIP_USERNAME DESTINATION: Dial a destination and enable the tester to control the call flow.
   • broadcast SIP_USERNAME DESTINATION /path/to/audio.wav: Dial a destination and play an audio file.
   • tts /path/to/the/audio "Some voice announcement": Use text-to-speech for announcements.

Use play and broadcast to test RTP streams and audio quality or to assess call recording functionality by capturing SIP and RTP dumps with the sngrep utility and analyzing them with Wireshark.

IMPORTANT: Audio files used by play and broadcast must be converted to a compatible format. The tts script handles this, or you can use the ffmpeg utility (also within the Docker container):

ffmpeg -y -i SOURCE.wav -ar 8000 -ac 1 -ab 64K CONVERTED.wav

When an inbound call reaches the conference and the user on SIP/0001 answers, the /recordings/line_1.wav announcement will be played to the calling party:

docker-compose run --rm --service-ports pjsua play 0001 /recordings/line_1.wav

Dial 00491234567890 from the trunk, and as soon as Asterisk answers, start playing the announcement /recordings/trunk.wav:

docker-compose run --rm --service-ports pjsua broadcast trunk 00491234567890 /recordings/trunk.wav

Use both cases to test call transfers (broadcast as a trunk, receive the call as agent 0003, and transfer it to 0001).

Configuration settings are in the .env file. IMPORTANT: Except for SIP_USERNAME, it shares the same configuration options. There are two ways to provide different credentials, such as a SIP password:

• Copy the whole folder and use a different .env file.

• Use the -e KEY=VAL argument with docker-compose:

  docker-compose run --rm --service-ports -e SIP_PASSWORD=foobar \
                     pjsua play 0001 /recordings/line_1.wav

This test tool is based on Alpine Linux and runs in a Docker container. To troubleshoot, enter the container using:

docker-compose run --rm --service-ports --entrypoint=sh pjsua

You can add necessary tools such as vim or sngrep using:

apk add sngrep mc vim