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_________________________________________________________________________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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The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
Build your next product on our scalable cloud platform.
Join our online community to chat in real time https://signalwire.community
Professional FreeSWITCH Services
sales@freeswitch.com
https://freeswitch.com
Official FreeSWITCH Sites
https://freeswitch.com/oss
https://freeswitch.org/confluence
https://cluecon.com
FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org
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