webrtcbin get-stats don't contain any rtpstats.

[Faraz Khan]

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
[hidden email]
https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel

[Matthew Waters]

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
[hidden email]
https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel

_______________________________________________
gstreamer-devel mailing list
[hidden email]
https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel

[Faraz Khan]

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 <[hidden email]> 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
[hidden email]
https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel

_______________________________________________
gstreamer-devel mailing list
[hidden email]
https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel

[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 <[hidden email]> 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
[hidden email]
https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel

_______________________________________________
gstreamer-devel mailing list
[hidden email]
https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel

[Faraz Khan]

Gotcha. This issue made me think otherwise. Seems like these guys (gstreamer devs) are getting way more stats than me:

https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/889

On Tue, Oct 6, 2020, 11:29 PM Matthew Waters <[hidden email]> wrote:

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 <[hidden email]> 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
[hidden email]
https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel

_______________________________________________
gstreamer-devel mailing list
[hidden email]
https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel

[Faraz Khan]

For posterity the problem was that I was using gstreamer 1.16 where this stuff isn't linked up. All works fine using master. Thanks for the webrtcbin folks! 

On Wed, Oct 7, 2020 at 7:30 AM Faraz Khan <[hidden email]> wrote:

Gotcha. This issue made me think otherwise. Seems like these guys (gstreamer devs) are getting way more stats than me:

https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/889

On Tue, Oct 6, 2020, 11:29 PM Matthew Waters <[hidden email]> wrote:

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 <[hidden email]> 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
[hidden email]
https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel

_______________________________________________
gstreamer-devel mailing list
[hidden email]
https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel