Comments (5)
mid
corresponds to a media stream, not a participant. Your publisher is sending both audio and video, so you get two different mids. Maybe you're confused with msid
. That said, please post questions like this on the group, we only use GitHub for code issues.
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@atoppi made me notice that you may be referring to the fact both are video streams, and that both refer to the same source, that is feed_mid: "0"
. In theory, this can happen if:
- the publisher is sending video only
- you subscribed to that publisher once
- you subscribe to that publisher again
In that case, it would be normal for you go get the same feed twice. If not, there may be a bug. Make sure (via debugging the client side) that you're not unknowingly sending the same subscribe request twice (e.g., once when doing join
, and once now via subscribe
). As such, please provide the full API sequence, and possibly the SDPs involved (ideally as a gist or pastebin link, not inline here).
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Yes, both are video streams. We don't use audio of videoroom, for audio we use audiobridge.
We are collecting the debug log for both server and client.
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Here is the gist URL
providing both server and client log.
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Your log demonstrates what I thought: you're subscribing to the same user twice yourself. Before calling subscribe
for ID 7628034449938680
, you also used switch
for the same feed. As such, it's a problem in your client side login, not Janus. Closing.
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Related Issues (20)
- [1.x] Sip session get stuck HOT 12
- No call back triggering in 'onlocalstream' HOT 2
- [1.x] Streaming plugin does not send "started" event to event handlers HOT 2
- Add unauthenticated no-payload ping endpoint for health checks HOT 3
- Build fails with latest libsrtp version HOT 9
- [0.x] [janus.js] black screen or streaming stops periodically when connected through WebRTC (Chromium only) HOT 4
- [1.x] Issue with outbound calls for SIP Gateway HOT 3
- [1.x] videoroom: remote publisher doesn't release RTP/RTCP ports HOT 10
- [1.x] Does AudioBridge supports multiple participant join from RTP in a single room . Getting `Already in a room` Error for 2nd participant HOT 3
- [1.x] rtp_port_range not being respected HOT 2
- [1.x] Crash / segfault on 1.2.2 HOT 5
- [1.x] system under heavy load stops to handle calls wia ws api HOT 16
- [1.x] Crash / segfault on 1.2.3 HOT 10
- [1.x] inconsistency happen when more than 6 users join at same time HOT 2
- [1.x]inconsistency happen when more than 6 users join at same time with different browsers (web and mobile) HOT 8
- [1.x] videoroom: support of string_ids/string_ids_user HOT 2
- [1.x] Increased event handlers SIP logging buffer size one more time or make it configurable HOT 5
- [1.x] build failure on macos HOT 3
- [0.x] Possible memory leaks found with libasan latest HOT 12
- [1.x] SIP plugin "Couldn't connect audio RTP? -- 22 (Invalid argument)" HOT 5
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