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I had just reproduced following error with Janus version with the commit. It doesn't happen with the earlier build (a few months back one) I have.
https://github.com/meetecho/janus-gateway/commit/6…
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In the janus_audiobridge plugin, when a user with specific user_id rejoin the pocroom with new session, the previous participant with same user id can be found in audiobridge->participants.
When the …
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I've been investigating managing simultaneous SIP calls. It seems reasonable to attach another SIP plugin as soon as you're on an active call to be able to receive incoming call requests separately du…
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SIP audio stopped working in v0.7.4, specifically starting with commit dca6fec91c8c4f1da096bbad295a7725a04c5f00.
This is the line we believe to be causing problems for us:
https://github.com/meete…
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Help needed!! I have managed to successfully initiate connection (register, call, accept) between the peers but cannot get the remote streams showing yet the local stream is showing.
Here is the …
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Not easy to reproduce (only showed up during load testing):
```
ASAN:DEADLYSIGNAL
=================================================================
==13==ERROR: AddressSanitizer: SEGV on unknown a…
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Amazing work guys, really. I'm super excited by this new webRTC integration! Anyways, I've deployed Gstreamer Master onto a Raspberry Pi and got the sendrecv Python demo to work with the provided webs…
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@mdblack98 The recent changes on `get_ant` (or `y`) command break existing Hamlib clients that use the command, as it now expects an argument.
1) Based on the code, all I can see is that there is a…
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Hi,
I am using Janus Gateway to stream my video from Gstreamer to Html. It works fine in case of Firefox but it fails on Chrome with the following error:
`streamingtest.js:166 WebRTC error: DOM…
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Join in for the weekly Monero Research Lab meeting.
**When**: Wednesday, 29 April 2020 @ 17:00 UTC
**Where**: #monero-research-lab (freenode/matrix)
**Agenda**
1. Greetings
2. Roundtable
…