webrtcbin get-stats don't contain any rtpstats.

[Matthew Waters]

No, you don't 'need' to do anything for rtpbin to expose the stats, they
just aren't always exposed in the webrtcbin stats as the webrtcbin stats
are incomplete.

Cheers
-Matt

On 7/10/20 5:06 pm, Faraz Khan wrote:
> Oh interesting. Seemed like rtpinputstats (from the RR reports) were
> already baked in. I'm specifically looking for fraction lost and
> interarrival jitter. Do I have to install a signal for rtcp packets
> and extracts those reports myself? 
>
> For some reason I thought rtpbin would do this for me.
>
> Thanks!
>
> On Tue, Oct 6, 2020, 10:41 PM Matthew Waters <ystreet00 at gmail.com
> <mailto:ystreet00 at gmail.com>> wrote:
>
>     Various information provided by the get-stats signal are
>     incomplete.  You may have to implement some of the information you
>     need inside webrtcbin itself.
>
>     Cheers
>     -Matt
>
>     On 7/10/20 3:25 pm, Faraz Khan wrote:
>>     Hi everyone,
>>     Gstreamer noob here - but have worked on C/C++ audio/video apps
>>     for many years now. I have successfully established a webrtc
>>     connection with electron as the receiver (and gstreamer as
>>     send-only) and am now looking to implement some sort of
>>     simplistic bitrate control and also just to understand how
>>     gstreamer works.
>>
>>     My first attempt was to get webrtc peerconnection stats. I did:
>>
>>     GstPromise *p = gst_promise_new_with_change_func(_on_stats, NULL,
>>     NULL);
>>     g_signal_emit_by_name(peer->webrtcbin, "get-stats", NULL, p);
>>
>>     And decode them using:
>>
>>     gboolean on_stats_iter(GQuark field, const GValue * value,
>>     gpointer pfx) {
>>         gchar *str = gst_value_serialize (value);
>>         g_print ("%15s: %s", g_quark_to_string (field), str);
>>         std::cout << std::endl;
>>         g_free (str);
>>         return TRUE;
>>     }
>>
>>     static void _on_stats (GstPromise * promise, gpointer user_data) {
>>         const GstStructure *reply = gst_promise_get_reply(promise);
>>         gst_structure_foreach(reply, on_stats_iter, (gpointer)reply);
>>     }
>>
>>     And this generates: 
>>     peer-connection-stats: "peer-connection\,\
>>     data-channels-opened\=\(uint\)0\,\
>>     data-channels-closed\=\(uint\)0\,\
>>     data-channels-requested\=\(uint\)0\,\
>>     data-channels-accepted\=\(uint\)0\,\
>>     type\=\(GstWebRTCStatsType\)GST_WEBRTC_STATS_PEER_CONNECTION\,\
>>     timestamp\=\(double\)180230225.79899999\,\
>>     id\=\(string\)peer-connection-stats\;"
>>     codec-stats-sink_0: "codec\,\
>>     type\=\(GstWebRTCStatsType\)GST_WEBRTC_STATS_CODEC\,\
>>     timestamp\=\(double\)180230225.79899999\,\
>>     id\=\(string\)codec-stats-sink_0\,\ payload-type\=\(uint\)96\,\
>>     clock-rate\=\(uint\)90000\,\ ssrc\=\(uint\)2002222005\;"
>>     transport-stats_webrtcdtlstransport0: "transport\,\
>>     type\=\(GstWebRTCStatsType\)GST_WEBRTC_STATS_TRANSPORT\,\
>>     timestamp\=\(double\)180230225.79899999\,\
>>     id\=\(string\)transport-stats_webrtcdtlstransport0\;"
>>     ice-candidate-pair_webrtcnicetransport0: "transport\,\
>>     type\=\(GstWebRTCStatsType\)GST_WEBRTC_STATS_TRANSPORT\,\
>>     timestamp\=\(double\)180230225.79899999\,\
>>     id\=\(string\)ice-candidate-pair_webrtcnicetransport0\;"
>>     codec-stats-sink_1: "codec\,\
>>     type\=\(GstWebRTCStatsType\)GST_WEBRTC_STATS_CODEC\,\
>>     timestamp\=\(double\)180230225.79899999\,\
>>     id\=\(string\)codec-stats-sink_1\,\ payload-type\=\(uint\)97\,\
>>     clock-rate\=\(uint\)48000\,\ ssrc\=\(uint\)1224562656\;"
>>
>>
>>     This has the correct SSRC and codec info - but missing RTP
>>     information and the data channel stats are incorrect (I do have a
>>     open and working data channel).
>>
>>     How does one proceed to debug this? What am I missing?
>>
>>
>>     _______________________________________________
>>     gstreamer-devel mailing list
>>     gstreamer-devel at lists.freedesktop.org <mailto:gstreamer-devel at lists.freedesktop.org>
>>     https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel
>

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