Comments (5)
please take a look here:
https://github.com/baresip/baresip/wiki/Guidelines-for-reporting-Issues
warnings in system headers is outside the scope of baresip.
please include a full build log.
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The spandsp.h
warnings should be fixed by: #2836
AVCaptureDeviceTypeExternalUnknown deprecation
needs another PR for replacement (maybe version based)
from baresip.
hi, could you please try again with Git HEAD ?
from baresip.
Can you try: #2854
from baresip.
Can you try: #2854
Hey! Sorry for the late reply, holidays season.
This PR bulids perfect, no UTF error, and no AVCapture errors. Runs perfect also.
But! The Audio is very distorted, when I call different clients/users. If I build from tag release/version "v3.7.0" I have no issues with the actullay audio streamed to me. I can capture it if you like.
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Related Issues (20)
- libbaresip cmake fails to find FFMPEG HOT 2
- Wrong UA_EVENT on incoming call with early video HOT 3
- Module aufile does not work properly as audio source for calls HOT 11
- ua: sip:alice@office: UA not found: HOT 5
- How to integrate baresip and ffmpeg?
- Echo cancellation HOT 4
- v3.9.0 build failure in Homebrew HOT 2
- Could NOT find RE (missing: RE_INCLUDE_DIR)
- Call getting rejected with "no common audio codecs - rejected" HOT 12
- -- Could NOT find GZRTP (missing: GZRTP_INCLUDE_DIR) HOT 4
- aufile: ITU G.711 mu-law with 8000 sample rate are played much too fast HOT 8
- calls from -e are starting before register HOT 7
- gst module not found in debian 11 HOT 4
- Assertion failure in 'echo' module. HOT 2
- webrtc demo cant display video on chrome 122.0.6261.58 version HOT 5
- 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
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