Crash in

[Barry Porch]

Hello,

I've developed a telephony app built on Electron and SIP.js, which connects to a remote webrtc <-> SIP gateway and streams audio via webrtc.  

I have a recurring issue with a crash of the entire Electron renderer process due to an error raised within the webrtc audio encoder.  The error received is:

# Fatal error in: ..\..\third_party\webrtc\modules\audio_coding\codecs\cng\audio_encoder_cng.cc, line 284

# last system error: 0

# Check failed: info.encoded_bytes == 0 (160 vs. 0)

# Encoder delivered data too early.

The issue occurs when a caller is rapidly reinvited on the far side, such as when a call is redirected between voicemail to an ACD path and back to voicemail.  The issue appears timing dependent and will happen about 1 out of 3 times under these conditions.  

I've implemented an ugly workaround of disconnecting the remote stream from the local media element as soon as I get notified of the invite, and then reconnecting the stream 1.75 seconds later.  This avoids the crash but then rapidly plays back the buffered audio which is not a good user experience.

Any help or suggestions would be greatly appreciated.

Barry

[Natalie Silvanovich]

I'd recommend you file a bug for this in the tracker: https://bugs.chromium.org/p/webrtc/

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[Philipp Hancke]

This was mentioned in https://groups.google.com/g/discuss-webrtc/c/SdVkVgXHBvg/m/2cB3pnBaAQAJ but no solution.

To view this discussion on the web visit https://groups.google.com/d/msgid/discuss-webrtc/CABVaO-7pj9eSFG%2BS7KcfdakXdoZXr6P34OvR%2Bf-UFLZMteNuXg%40mail.gmail.com.

[Jon Arne Grødal]

FYI: A fix for the audio_encoder_cng.cc, line 284 crash has now been submitted:
https://issues.webrtc.org/issues/42225071

I have tested this in Chrome Canary, and I no longer get this crash after reinvites.