Comments (2)
@pedrokiefer. I guess you where using the master branch of jssip-demos. I've just updated the repo. Please retry.
Regardless 'a=group BOUNLDE audio video' line in sdp, Asterisk won't complain if properly configured (https://wiki.asterisk.org/wiki/display/AST/Asterisk+WebRTC+Support). Keep the 'enable video' checkbox untagged.
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@jmillan Great! now the call completes, but I don't get any audio through, it just disconnects 30 seconds latter. I think there is something definitely wrong with my asterisk configuration... although I'm using all the configs recommended on asterisk's wiki.
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Related Issues (20)
- Abnormal session cancellation method under race condition
- Type definition errors in 3.10.0 HOT 1
- Error when getting video track status on Firefox
- Play audio or tone to remote user. HOT 2
- The new JsSIP.UA(configuration).call method does not return the status after execution
- import JsSIP from "jssip"; 打印JsSIP undefined HOT 2
- Error 'setLocalDescription' on Chrome HOT 1
- BYE: Request-URI does not point to us HOT 34
- Incorrect SDP a=sendrecv in firefox HOT 7
- I want to send and Receive message using JSSIP
- Sending a silent stream on latest Chrome HOT 1
- Error when trying to change the microphone during a call HOT 1
- Add custom header JsSIP - Asterisk HOT 1
- JsSIP processing re-INVITE HOT 1
- SDP is cut off between recieving 200 OK and the sdp event being emitted HOT 2
- On iOS, navigator.mediaDevices is `undefined` unless page is served in a secure context, but Errors are swallowed
- Why is there a 2-second delay when exiting? According to the source code, it seems to have no practical significance. HOT 3
- Codec Preference Selection Not Supported in JsSIP HOT 4
- "User denied media access"
- Application Build with Angular 18 fails due to typing errors HOT 2
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