Comments (8)
Is audio level RTP extension being negotiated?
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Not only that. The room must have been created with support for the extension, and the related properties must be tweaked to use good values for detecting users talking. By default this is disabled, and the default properties are probably not sensitive enough for most use cases (meaning no speech could ever be detected). In both cases, this is not an issue (so out of scope here) but a matter of properly configuring rooms and participants.
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The room is configured correctly, and other devices, such as the Android app, are receiving Talking events. I am using the desktop version of our WebRTC client with the SIPSorcery WebRTC library.
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Is audio level RTP extension being negotiated?
Would you please explain a bit.
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The room is configured correctly, and other devices, such as the Android app, are receiving Talking events. I am using the desktop version of our WebRTC client with the SIPSorcery WebRTC library.
Janus has no idea if the client is a mobile endpoint or something else: if the feature is available, it will be used. As Alessandro pointed out, though, this will only work if the audio-level extension is negotiated by the WebRTC endpoint, as we use that to detect voice activity. You should check iif SIP Sorcery does negotiate them in the SDP: if not, you have your answer.
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@mail2mhossain have you checked if SIP Sorcery negotiates the audio-levels RTP extension, and if the extension is indeed present in RTP packets?
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In Android SDP, there are audio-level RTP mappings:
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
However, this mapping is missing in the desktop SDP.
I have manually added it to the desktop SDP, but I'm still not receiving the talking events.
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That's because, as I said in my previous messages, the extension must be negotiated but also present in RTP packets. Closing as not an issue.
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Related Issues (20)
- [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 19
- [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 10
- [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 7
- [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
- [1.x] Transport Websockets plugin binds to ANY network interface HOT 4
- Failed to compile janus on an arm machine inside docker container with ubuntu 24.04 as image HOT 1
- [1.x] Im used Janus in docker contaner with docker-compose HOT 1
- [1.x] Commandline option "--version" no longer exists HOT 1
- [1.x] why is it recommended to use libnice v0.1.18? libnice 0.1.19 exists a number of bug fixes HOT 1
- [1.x] Sip plugin crash on Options HOT 6
- [1.x] HOT 1
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