Softphone, webphone and VoIP server Forum

HomeHomeDiscussionsDiscussionsGeneralGeneralWebRTC 2 SIP audio problemWebRTC 2 SIP audio problem
Previous
 
Next
New Post
9/7/2016 8:57 AM
 
Hi,

I've been testing your webphone solution, and it works perfectly with your online webrtc2sip gateway. Now i've tried to implement the gateway on my own server, and while i get the webphone to connect to my VoIP server (3cx), i'm getting no audio on phone calls. Also, when i disconnect the call on the other end, the webphone reports it's still in a call. Any ideas on what i might be doing wrong?
Everything is set up on a local lan, and firewall is disabled on webrtc2sip server.

Another thing i noticed:
C:\Users\Administrator>netstat -an | findstr :506
  TCP    192.168.30.26:5060     0.0.0.0:0              LISTENING
  TCP    192.168.30.26:5061     0.0.0.0:0              LISTENING
  UDP    192.168.30.26:5060     *:*

It looks like it's not listening on UDP. Any idea why?
 
New Post
9/9/2016 7:14 AM
 
If the webphone is using WebRTC then it is normal to not listen on UDP because WebRTC signaling goes via https/websocket.

You might also check with some third-party WebRTC client to see if that works, for example SIPML5:
https://www.doubango.org/sipml5/expert.htm
https://www.doubango.org/sipml5/call.htm

If this is working, but our webphone doesn't work, please send us a log from the browser console and we will check it immediately.
 
New Post
9/9/2016 9:53 AM
 
I've tested sipml5, and i'm getting exactly the same behaviour as with your webphone. It connects to the server, starts a call, my phone rings, but there's no audio when i answer, and when i hang up on the phone softphone says it's still in call.

I've received some new public ip's today so i'll try to move entire testing solution there and see how it goes. In the meantime, i'm guessing i've set up something wrong in webrtc gateway.
 
New Post
9/9/2016 10:52 AM
 
Ok, but this proves that the problem is with your server software (or it's configuration) and not with our webphone.

If you wish, we can provide you a WebRTC SIP gateway software for free to be used with 3CX which will handle all this much better then the built-in 3CX webrtc module. (We can provide the exact same software what is used for our public WebRTC gateway).
 
New Post
9/9/2016 11:23 AM
 
I'm not using 3cx webrtc. It doesnt actuqally work for what i need it to. I have allready installed your free WebRTC gateway and that's the one i'm having problems with.
 
New Post
9/10/2016 3:33 PM
 
Ahh, I understand the problem now.
There is an issue with the latest release as google have just changed the DTLS SSL support. We already updated our public gateway, but haven't updated the downloadable software.

Please send us (to support@mizu-voip.com) a remote access to your server and we will apply the necessary patches.

Otherwise we are going to release a new version soon with this fixed.
 
New Post
10/19/2016 5:54 AM
 
I just want to add here that now this issue is fixed also in the public downloadable package: free webrtc sip gateway
 
Previous
 
Next
HomeHomeDiscussionsDiscussionsGeneralGeneralWebRTC 2 SIP audio problemWebRTC 2 SIP audio problem


You need to register / login to be able to post in the forum