vicidial.org

Hi all

recently I have had problems with the webrtc and been investigating which is the reason.I have a cluster with 7 dialers which has been working very well for a year, the servers are hosted in aws.

VERSION: 2.14-773a
BUILD: 201101-1056
Version: 2.14b0.5
SVN Version: 3309
DB Schema Version: 1609
DB Schema Update Date: 2020-11-03 21:32:28

A couple of days ago several ext lost audio and I realized that all the ext of 2 of my dialers had the same problem , i double check conf but all are ok and no recent changes were made.
I noticed that if I connect to the VPN, where my servers are, the audio works without a problem, but not from the internet.I also noticed that if I restart the server several times at some point I recover the audio or when restoring the vps. I thought it was a problem with the dialers that presented the failure, but today other dialers also failed. i reboot several tiems and it works again

i check serverconf , firewall ports and all is ok .
I have in sip :
nat=force_rport_comedia

and rtp.conf :

icesupport=true

stunaddr=stun.l.google.com:19302







I

It's aws. Switch

apparently the problem is the stun server , in the traces, the external ip of my server appears as 0.0.0.0 I am trying to force placing the public ip in the rtp so as not to have to use the stun server

the Switch?

put this in rtp.conf work , but i dont undertand why the stun server stop working

The following will include BOTH 192.168.1.10 and 1.2.3.4 during ICE
; negotiation instead of replacing 192.168.1.10. This can make it easier
; to serve both local and remote clients.
;

carpenox wrote:It's aws. Switch

Why the aws switch is the problem ?

AWS constantly makes changes on their backend that messes with webrtc

Ok thanks for the reply

Switch your stun server. Google stun died

carpenox wrote:Switch your stun server. Google stun died

does viciphone uses google stuns?

yes, edit the rtp.conf file, its the last line. heres a list of public stun servers: https://gist.github.com/mondain/b0ec1cf5f60ae726202e

I have created a quick article for anyone having this problem:

https://cyburdial.net/why-doesnt-my-web ... k-anymore/

I'm experiencing this issue in viciphone 2, Suddenly customer cannot hear the agent, but agent hears customer. In recording both side are recorded. but sometimes it is working fine both side can communicate. My stunaddr in my /etc/asterisk/rtp.conf is currently commented, should i edit this?

yes add a stun server

yes add a stun server

Thank you, how do I know that it is applied? Is there any command to show to the cli or logs?

just run a "core reload" in asterisk