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.js
On 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 * VoIP Engineer @ Quobis <http://www.quobis.com/> | e: sergio.garcia@quobis.com | t: +34902999465
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-- *Sergio García Ramos * VoIP Engineer @ Quobis <http://www.quobis.com/> | e: sergio.garcia@quobis.com | t: +34902999465