Not sure if there are any folks on the list that can help, but here are some logs. I have poured over them and can't find anything different in either the console logs or freeswitch logs across the two versions of FireFox. [One Way Audio - Failure] https://nofile.io/f/VnF5cFGFseg/freeswitch-log-failure-FF61.log https://nofile.io/f/yjT6OCfAP7w/console-log-failure-FF61.txt [Two Way Audio - Success] https://nofile.io/f/HVXzCmE1VVx/freeswitch-log-success-FF60.log https://nofile.io/f/GzyJrSjs7aS/console-log-success-FF60.txt The call flow we have in this example is that a normal SIP over UDP call comes into FreeSwitch using G.711 codec. Freeswitch invites to the WSS client (using Opus codec) and bridges the media. In the Freeswitch logs, you will see the ingress INVITE as well as the egress INVITE to the browser. On the "failure" calls - there is no audible media being delivered, but we are seeing RTP packets using tcpdump, so we don't have a NAT/Firewall issue or anything. On Thu, Jun 28, 2018 at 3:02 PM Geoff Mina <gmina@connectfirst.com> wrote:
Is anyone else having issues with WebRTC and the latest release of FireFox? We are showing one way media - with media flowing properly TO the browser, but no media coming back from the browser.
Everything in the logs looks OK - no errors. We are using SIP.js as the library of choice for signaling.
We are showing SRTP packets flowing bi-directionally (using tcpdump), but since they are DTLS encrypted, I am unable to see exactly what is in each packet to determine if there is media being sent from the browser or they are just empty packets.
Any ideas and/or other folks seeing the same issue - would love to hear your thoughts.
Thanks, Geoff