Hi Chris,I'm not using JSSIP or Sipml5, I'm using our own solution. As I said in the previous email, we are sending IP 0.0.0.0, port 0 and media inactive. It seems FS doesn't like this and I would like to know the reason.I've changed our end to send a=sendonly (instead of a=inactive) and to not change the port to 0, and FS accepted it. The problem now is that after one second hearing the music on hold it stops, and I can never get the media back (I'm using TURN server to avoid NAT/Firewall problems).Thank you.2015-07-23 18:11 GMT+02:00 Chris Tunbridge <blasterjr@gmail.com>:Sergio, please fill in some additional information like which JS WebSocket client you're using, there are lots of hold issues with SipML5 and JsSIP, the only one that i know that has working hold music is sip.js (was a fork of JsSIP) and I've personally contributed towards making the hold function correctly on sip.jsOn Thu, Jul 23, 2015 at 5:17 AM, Sergio García <sergio.garcia@quobis.com> wrote:_________________________________________________________________________Hello all,I am using FreeSwitch as a WebRTC gateway thanks to its Websocket support (in my case WSS), but I'm facing this strange problem. Audio and Video calls are working perfectly fine, but when I try to set a call On Hold, FreeSwitch replies with "488 Not Acceptable Here" error to the ReINVITE I'm sending.The only error I can see in the logs is:[ERR] sofia.c:7280 Reinvite Codec Error!The only difference between the original INVITE and this ReINVITE is that I try to set IP address to 0.0.0.0, port to 0 and media attribute to inactive. I don't understand what part of the SDP, FS doesn't "like".Attached you can find a more detailed log file.Thank you very much in advance.Regards,--Sergio García Ramos
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