Got freezed frame during connection between a SIP endpoint and JSSIP demo site
Reported by
antira...@gmail.com,
Aug 30 2017
|
|||||
Issue descriptionChrome Version : 60.0.3112.113 OS version : Windows 7 Service Pack 1 Network : Cable Audio/Video format : Audio: G.722 , Video: H.264 Special chrome flags : ---enable-logging --vmodule=*/webrtc/*=1 --ignore-certificate-errors --flag-switches-begin --flag-switches-end chrome Video issue, Audio issue, both, neither? Video Flash or HTML5? HTML5 What steps will reproduce the problem? (1) use JSSIP (https://tryit.jssip.net/) and registered it to a Kamailio proxy server (2) make another classical SIP client also register to the server (3) take turns making calls each other several times (Until now I cannot find the exact reproducible steps) What is the expected result? - the media streams just work normally. What is the actual result? - some times after call started for seconds, JSSIP side got a frozen frame and not recover unless you hangup it and start a new one. Any additional information (anything else which may help us debug the issue)? - I checked whether JSSIP side did receive the video SRTP stream or not by capturing the packets, but the packets showed that the browser indeed obtained the media contents, then I just wonder if it's the issue of the decoder the browser used. log files as attached. Please inform me if any other information is need. Thanks. :)
,
Sep 12 2017
,
Sep 13 2017
,
Sep 14 2017
,
Sep 14 2017
antirazin: Would you be able to submit a screenshot of the graphs in chrome://webrtc-internals when you have the freeze? In the supplied log, there are 122 log lines like this: [5444:21988:0830/145326.239:WARNING:rtp_rtcp_impl.cc(175)] webrtc::ModuleRtpRtcpImpl::Process: Timeout: No increase in RTCP RR extended highest sequence number. That indicates that the receiver stopped receiveing RTP packets. But if the logs are from the Chrome+JSSIP endpoint, that would indicate a freeze in the legacy SIP client. But IIUC, you see the freeze in Chrome?
,
Nov 15 2017
Not enough information for us to investigate further, and it indicates that this could be an issue with a third party sip client. Closing. |
|||||
►
Sign in to add a comment |
|||||
Comment 1 by hdodda@chromium.org
, Sep 5 2017Labels: Needs-Triage-M60 TE-NeedsTriageHelp