WebRTC/SipJS No Incoming Audio

[taylorman57]

I have a browser-based WebRTC softphone setup using SipJS and FusionPBX. My FreeSwitch version is 1.10.2

If I call out from SipJS to an internal extension or an external pstn number, audio is working both ways.
If I call in to SipJS from a SIP softphone or and external pstn call then the SipJS client cannot hear anything (but the other end can)
If I make a SipJS to SipJS call, audio works both ways
SIP Softphone to Sip Softphone (grandstream app or Zoiper) works fine.

The server is on a public IP. I have tried incoming calls from behind a NAT and not behind a NAT.
SipJS client is using a google STUN server

There is no difference when using Firefox or Chrome.

I currently have the following ports open:
TCP: 22, 80, 443, 2855-2856, 3478-3479, 5066, 8081-8082
TCP/UDP: 5060, 5061, 5070, 5080, 5081, 7443
UDP: 16384-32768

I saw in the sipjs log some ports higher up, so I temporarily opened 45000-65500 just to see if that helped but it did not make a difference.

I have my ip address set in the variables and sip profiles and have followed every other guide/forum post I can find.

I have attached logs from both freeswitch cli and the sipjs console log.

Does anyone on here have any ideas on how to get the incoming audio stream to work or have some troubleshooting tips so I can trace the problem?

Part of me is leaning towards it being a codec issue, seems how it happens when I go from WebRTC to SIP, but when calling out I don't have the problem so that makes me think it is a firewall/routing issue.

 

[KonradSC]

I saw an issue like that with an older version (0.7.8) of sipjs that ctxsip used. This guy made a bunch of changes to allow for the latest version (0.15) and submitted a pull request.
https://github.com/collecttix/ctxSip/pull/36

I took that guy's pull request and made a few tweaks to 0.15 to get mute and hold working. https://github.com/konradSC/Fusionpbx-Public/blob/master/Apps/webphone/phone/scripts/SIP.js/sip.js

 

[taylorman57]

I can confirm your implementation does work. I was actually using a fork of ctxsip that I had done my own modifications to in order to make DTMF work. It looks like you have also fixed that.

I am working on a desktop app sort of like the vonage/ring central ones that can do the phone calls, texting, and visual voicemail. I'll share it on this thread when it is done.

Thank you so much.

 

[mark]

Its been a while since this post has been active. I implemented this on a second server. But it seems to be having registration failures between calls. Any idea why that would be?

 

[taylorman57]

Have you tried a regular sip softphone client to see if it works? That would rule out if it is a sip issue or a webrtc issue.