How to increase volume user when call by WebRtc?

configuration
freepbx
Tags: #<Tag:0x00007fafc91da120> #<Tag:0x00007fafc91d9f40>

(Iuiuadu9x) #1

I’m using freepbx distro.

Freepbx: 15.0.16.44
Current Asterisk Version: 16.4.1

I had build a web client using Webrtc ( with lib jssip.net) connect with FreePbx to call.

=====
When A using web client call phone number B.
B listen to A ok (clear)
But A listen to B is heard very small sound.

I was edited RG, TG in DAHDI Configuration to 30.0 after fwconsole restart but has not changed much.

My code add audio in stream to A can hear from B.

           var sipAudio = new Audio();
            session = ua.call(phone_number, options);
            session.connection.onaddstream = function (e) {
                sipAudio.srcObject = e.stream;
                sipAudio.volume = 1.0;
                sipAudio.play();
            };

So that, how to increase volume and what do config that?

Thank you.


(Game Bài Đổi Thưởng) #2

I don’t know about this issue, maybe someone will help you.


(Jared K Smith) #3

Changing the gain in DAHDI isn’t going to help with WebRTC calls – that would only affect analog phone calls coming in on an analog card.

With WebRTC and JSSIP, you’re really going to want to adjust the audio at your endpoints, not within Asterisk itself.


(Iuiuadu9x) #4

Thank you. But my “your endpoints” is B. B is user normal using mobile signal GSM (without internet).
Before call to destination B, FreePBX connect to Trunk device Yeastar TG1600. I also edit TG, RG but has not changed much.

Do you have another recommend for me?


(Iuiuadu9x) #5

You are right. My problem was resolved. I had changed TG, TX in my device TG1600 Yeastar before call to user is ok.
But must change TG, TX more than UI interface by command into SSH.