[Bug 751444] New: RTP: Keep track of all estimated round trip times observed per each report block

[GStreamer (GNOME Bugzilla)]

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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