[asterisk-bugs] [JIRA] (ASTERISK-27826) No sound in some calls (webrtc)
Asterisk Team (JIRA)
noreply at issues.asterisk.org
Wed Apr 25 21:47:51 CDT 2018
[ https://issues.asterisk.org/jira/browse/ASTERISK-27826?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]
Asterisk Team updated ASTERISK-27826:
-------------------------------------
Status: Waiting for Feedback (was: Waiting for Feedback)
> 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
> Assignee: Asterisk Team
> Labels: webrtc
>
> 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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