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gatheringTimeout Missing v3.x #432
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We don't want to make or maintain WebRTC workarounds. My personal goal would be to completely decouple JsSIP from the WebRTC/PeerConnection stuff, by providing an API that lets the application join both (see #427). I will let this issue open for now. If it takes too long before we do #427, we may add a setting for this need. |
It will be done when we get time for it within our professional priorities. Money would help prioritizing it. |
anything update for this issue?
|
as i updated to 3.2.5 so i made patch like this: `
` |
There is a new RTCSession event: icecandidate, which provides a mechanism to skip the remaining local ICE gathering process whenever you like. Whenever you are about to make an outgoing call or answer an incoming call, set the event at your will and tell the Session within the event handler whether you have enough candidates by executing the NOTE: If you don't need the same behaviour for the whole session, do register/unregister the event handler from the event conveniently. |
@jmillan can you please provide sample code to |
Sorry, how to do it is documented in the API. You must rely on the API and not on code snippets. |
@ibc totally understand that. I have tried to understand API, but could not figure out how to use it. I have been using https://sipjs.com/api/0.7.0/ua_configuration_parameters/ and it had straight forward setting |
Honestly I don't care how sipjs does it. We expose an API for it and it works if property used. |
@ibc okay, thanks |
Hi
This not a bug but a specific feature missed out while moving to adapter.js
We updated our JsSIP version to v3.0.0 everything works as expected but not 'gatheringTimeout'.
gatheringTimeout
is added back so users can pass it viapcConfig
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