[asterisk-bugs] [JIRA] (ASTERISK-27826) No sound in some calls (webrtc)

[Mikhail Ivanov (JIRA)]

Mikhail Ivanov created ASTERISK-27826:
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             Summary: No sound in some calls (webrtc)
                 Key: ASTERISK-27826
                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-27826
             Project: Asterisk
          Issue Type: Bug
      Security Level: None
          Components: Resources/res_rtp_asterisk
    Affects Versions: 15.3.0
            Reporter: Mikhail Ivanov


I have a problem with incoming (may be with outgoing too, not sure) calls to WebRTC clients (based on jssip.net library)

Sometimes (2-5% of all incoming calls) I have no sound (on both sides) on incoming calls.

RTP is going fine in both sides (local network)

If I turn on mixMonitor on Asterisk, I can see only noise in call (looks like a problem with srtp keys, but not sure)

https://www.dropbox.com/s/41nmwqhg0chcwl7/cf626000ac4601445d6cee3cd909188d.mp3?dl=1

Asterisk 15.3.0, JsSIP 3.2.8, tested in Chrome, Chromium and Firefox

If I turn off rtp encryption 

webrtc = no 
rtcp_mux = yes 
use_avpf = yes 
ice_support = yes 
media_encryption = no

and 

--disable-webrtc-encryption in crome

everything is fine, yes, it's workaround but not a solution



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