Comments (12)
You can start baresip with the -s
option. (edit: and maybe the -v
option, although for now we just need the traces)
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Thanks. I don't have access to system now. I will share a few hours later.
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Thanks, I will have a look.
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@maximilianfridrich, wow 🚀 👍🏽 .
That did solve the issue.
But what is interesting is that media encryption is turned off in Linphone configs.
Maybe it is an issue on their side then....?
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Sure got it. Thanks a lot for the help!
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@haricane8133 It is difficult to figure out what happened without knowing what was sent in the SDP offer.
Could you provide a trace of what was sent in the INVITE to BareSIP?
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Sure. Is there a way to turn on Verbose logs from Baresip, or should I intercept network?
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@maximilianfridrich, here are the logs after starting baresip
with -s
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Here are the logs after starting baresip
with -v
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It's a configuration issue. I was able to reproduce your problem, then I configured BareSIP differently and now it works:
In the global baresip configuration I loaded
module dtls_srtp.so
and to my account configuration I added
;mediaenc=dtls_srtp
Please let me know if that resolves your issue.
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Considering that our instance may get calls from multiple sources which may or may not have media encryption, or different media encryptions each, is it possible to configure our BareSip instance to support more than one at a time?
If we add the media encryption to our account configs as mentioned, would incoming calls without media encryption fail?
(I am unable to test this as of the moment as I am unable to determine what the Linphone app does)
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Maybe it is an issue on their side then....?
The incoming INVITE looks like it came from an Asterisk server. Maybe that Asterisk needs to be configured differently.
If we add the media encryption to our account configs as mentioned, would incoming calls without media encryption fail?
I'm honestly not sure, trying it out is probably best.
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Related Issues (20)
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- Writing custom modules to interface with baresip?
- Can not receive method NOTIFY when connecting to wss server HOT 2
- Multiple outgoing calls, only sending audio when one is accepted HOT 8
- May I ask how to set the sampling rate of saved audio files to 16000, with a default of 8000
- answered call has video even when not asked for HOT 18
- Thread Sanitizer warning for `test_call_tcp` for the stream->tx object HOT 2
- inreq_allowed is missing from Wiki and if not allowed produced incorrect negative response HOT 10
- Baresip gets stuck related to "audio: rtp timestamp wraps backwards" since 3.9.0 HOT 6
- Add ;inreq_allowed to addr-params
- Restore backwards compatibility by accepting MESSAGE requests by default
- netroam module HOT 2
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