Hi Mahmood, Make sure DTLS/ICE works. Check with tcpdump/wireshark packets are flowing between ends of the call. Define ICE servers to be used by JsSip. var options = { 'eventHandlers' : your eventHandlers, 'mediaConstraints' : { 'audio': true, 'video': true }, 'pcConfig' : { 'iceServers': [ {'urls': 'stun:stun.stunprotocol.org:3478'}, {'urls': 'stun:stun.l.google.com:19302'}, ] }, }; var session = ua.call(uri, options); kind regards, Piotr Piotr Gregor Software Engineer M: (+44) 07483 866 525 www: dataandsignal.com On Wed, Feb 12, 2020 at 4:07 PM Mahmood Alkhalil < mahmood.alkhalil@outlook.com> wrote:
Hi Everyone,
I've been trying to use JsSIp over Websockets, I'm able to register and i can confirm the registration is a success, but whenevr i try to make a call Freeswitch responds with SIP/2.0 488 Not Acceptable Here.
The JsSip API is showing SDP Failure, in Freeswitch logs it is telling that there is SDP/RTP matches but then it say no candidate found.
I'v attached the call trace, appreciate any help.
Thanks, Mahmood Alkhalil. _________________________________________________________________________
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