[Bug 751444] New: RTP: Keep track of all estimated round trip times observed per each report block
GStreamer (GNOME Bugzilla)
bugzilla at gnome.org
Wed Jun 24 07:09:50 PDT 2015
https://bugzilla.gnome.org/show_bug.cgi?id=751444
Bug ID: 751444
Summary: RTP: Keep track of all estimated round trip times
observed per each report block
Classification: Platform
Product: GStreamer
Version: git master
OS: Linux
Status: NEW
Severity: normal
Priority: Normal
Component: gst-plugins-good
Assignee: gstreamer-bugs at lists.freedesktop.org
Reporter: sancane.kurento at gmail.com
QA Contact: gstreamer-bugs at lists.freedesktop.org
GNOME version: ---
Created attachment 306004
--> https://bugzilla.gnome.org/attachment.cgi?id=306004&action=edit
enhance stats for RtpSources
RTSession only stores the last round-trip-time calculated per RB so it is being
continuously overwritten as long as a new ssrc is extracted from the
GstRTCPPacket. Further more, rtt stats are stored in the proper RtpSource (the
one wich we receive the report for), current implementation only stores it in
the sender stats which is weird so that any RTCP packet might contain several
RB belonging to different SSRCs.
On the other hand, a new GstStructure is provided with the RTPSource stats that
contains all observed SSRCs and their proper round-trip-time to them.
These stats are helpful to provide internal information about WebRTC statistics
such as defined in
http://www.w3.org/TR/webrtc-stats/#idl-def-RTCOutboundRTPStreamStats
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