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Issue 760465 link

Starred by 2 users

Issue metadata

Status: WontFix
Owner: ----
Closed: Nov 2017
Cc:
Components:
EstimatedDays: ----
NextAction: ----
OS: ----
Pri: 3
Type: Bug



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Got freezed frame during connection between a SIP endpoint and JSSIP demo site

Reported by antira...@gmail.com, Aug 30 2017

Issue description

Chrome 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. :)


 
webrtc_internals_dump_20170830.txt
634 KB View Download
Cc: hdodda@chromium.org
Labels: Needs-Triage-M60 TE-NeedsTriageHelp
@Adding TE-NeedsTriageHelp, as the TE doesn't have the required access to proxy servers.

Thanks!
Components: -Internals>Media Blink>WebRTC

Comment 3 by guidou@chromium.org, Sep 13 2017

Components: -Blink>WebRTC Blink>WebRTC>Network Blink>WebRTC>Video

Comment 4 by antira...@gmail.com, Sep 14 2017

chrome_debug_20170830.log
4.0 MB View Download
Labels: Needs-Feedback
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?

Comment 6 by holmer@chromium.org, Nov 15 2017

Status: WontFix (was: Unconfirmed)
Not enough information for us to investigate further, and it indicates that this could be an issue with a third party sip client. Closing.

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