We found a bug with Firefox 34 which causes not working calls from browser to SIP.<!–more–> The issue was caused by wrong media info in the WebRTC SDP from FF 34.
The issue was fixed in the build 1052. Please update your version to the latest one to get the bugfix.
Server builds 761-821 (9 May 2014 – 31 May 2014)
761 – Fixed DTLS problem AlertDescription.51
762 – Fixed some signaling issues related SIP timeouts.
764 – Fixed “light deadlock” issue in ICE
765 – Implemented sending of DTLS packets in one UDP packet.
771 – Added auto generation of DTLS private key and certificate.
776 – Optimized ICE candidate harvestr using setting enable_candidate_harvester=false.
Keep reflexive candidate only.
779 – Fixed XCAP respone hanlding for a large message.
788 – Added webcallserver script for starting and stopping WCS as a service.
The script ensures to start server using /etc/init.d webcallserver start and add the script to Linux autorun after reboot.
805 – Added setting client_mode=true which says server to use external IP for WebRTC and network interface API for SIP and RTP exchange with a SIP Proxy or media server.
818 – Minor fixes in the shutdown handling.
820 – Added ExtendedDailyRollingFileAppender to keep server logs for the given number of days.
821 – Added LESS_LOADED_NODE as a mode for load balancing and ‘randomize’ parameter for the FEWESTCALLS loadbalancing mode.
Client builds 491-497 (9 May 2014 – 31 May 2014)
491 – Fixed Xml message body parsing with IE.
492 – Added Notify JS layer about 183 Session Progress.
494 – Added Notify record complete.
496 – Fixed Disable stun server by default to speed up call initiation.
497 – Added setting suppress_ring_on_active_audio_stream=false
Here we had an issue with self-signed certificates used with Android Chrome browser for WebRTC calls
