Can't make webrtcbin connect to H.264 offering clients
Neil Young
foreverneilyoung at googlemail.com
Sun Aug 22 12:59:28 UTC 2021
Thanks for sharing. Will check it out. I'm aware of the MDNS candidates thing, but
a) I'm caring about those
b) My Chrome is disabled to send them
It is definitely not an ICE issue, I have
> Am 22.08.2021 um 14:57 schrieb Nicolas Dufresne <nicolas at ndufresne.ca>:
>
>
>
> Le dim. 22 août 2021 08 h 48, Neil Young <foreverneilyoung at googlemail.com <mailto:foreverneilyoung at googlemail.com>> a écrit :
> BTW: I will also be happy to hand out a POC app to everybody willing to help. You would only need to have a webrtcbin capable installation of GST and Python3. Works out of the box, no strings attached.
>
> An answering app, I didn't link it before cause it's demoware. See the known issue about mdns candidates though, this is a missing feature that can pause problem.
>
> https://gitlab.collabora.com/showcases/ibc-2019/multistream-server <https://gitlab.collabora.com/showcases/ibc-2019/multistream-server>
>
> In general, specially at this time of the year, a lot of folks are on vacation, leaving other really busy. Expecting someone to look into your issues for free in such a short timeframe is not realistic.
>
>
>> Am 22.08.2021 um 14:46 schrieb Neil Young <foreverneilyoung at googlemail.com <mailto:foreverneilyoung at googlemail.com>>:
>>
>> Well, I have published everything I did so far. I confirm, it works if webrtc/python is the offerer and it also works as described with an incoming offer, if the pipleline is using vp8 instead of H.264.
>>
>> But it does not no work with H.264. And unless anybody points me to the obvious mistake I have made with my handling there is nothing to be taken back with my conclusion.
>>
>> Prove me wrong, if you can. It doesn't help just to say "it works". There are very rare sample of answering apps with webrtcbin. And I'm doing nothing else then those.
>>
>>
>>
>>
>>> Am 22.08.2021 um 14:41 schrieb Nicolas Dufresne <nicolas at ndufresne.ca <mailto:nicolas at ndufresne.ca>>:
>>>
>>>
>>>
>>> Le dim. 22 août 2021 07 h 45, Neil Young via gstreamer-devel <gstreamer-devel at lists.freedesktop.org <mailto:gstreamer-devel at lists.freedesktop.org>> a écrit :
>>> Finally giving up on this. Python, webrtcbin and H.264 offer - this does not work
>>>
>>> Hi Neil, I'm sorry this did not work you and no one was available in your required delay to freely support you.
>>>
>>> This answer is simply to state that this works for others, the webrtcbin and rtpbin pipeline are known to be difficult, but not broken (except for few missing webrtc features of course).
>>>
>>>
>>>
>>>
>>>
>>>> Am 21.08.2021 um 20:23 schrieb Neil Young <foreverneilyoung at googlemail.com <mailto:foreverneilyoung at googlemail.com>>:
>>>>
>>>> I'm hanging on this for a week now, and it seems so easy, but whatever I do, I can't make it work
>>>>
>>>> My app is a Python GST app, running on 18.4. It supports two modes: "call" and "answer". In call mode the app immediately issues an H.264 or VP8 OFFER (depending on a setting). This works in 100 % of the cases.
>>>>
>>>> What also works is when the app ANSWERs a VP8 offer. Just H.264 OFFERS are not working, even though the SDP answer looks good. I just don't get the video to remote.
>>>>
>>>> My app works like so:
>>>>
>>>> 1) When the app has a connection to the signaling server it creates a webrtcbin object:
>>>>
>>>> def build_webrtcbin(self):
>>>> self.webrtcbin = Gst.ElementFactory.make("webrtcbin")
>>>> self.webrtcbin.set_property("bundle-policy", "max-bundle")
>>>> self.webrtcbin.set_property("stun-server", "stun://stun.l.google.com:19302 <>")
>>>> self.webrtcbin.connect('on-negotiation-needed', self.on_negotiation_needed)
>>>> self.webrtcbin.connect('on-ice-candidate', self.on_ice_candidate)
>>>> self.webrtcbin.connect('pad-added', self.on_pad_added)
>>>> self.webrtcbin.connect('notify::connection-state', self.on_connection_state)
>>>> self.webrtcbin.connect('notify::signaling-state', self.on_signaling_state)
>>>> self.webrtcbin.connect('notify::ice-connection-state', self.on_ice_connection_state)
>>>> self.webrtcbin.connect('notify::ice-gathering-state', self.on_ice_gathering_state)
>>>>
>>>> self.pipeline = Gst.Pipeline.new()
>>>> self.pipeline.add(self.webrtcbin)
>>>>
>>>> Nothing special I guess.
>>>>
>>>> 2) When the SDP OFFER comes in, the app does this:
>>>>
>>>> 2a) It starts the H.264 pipeline with `start_pipeline`.
>>>> 2b) It emits the remote description to the local webrtcbin
>>>>
>>>> self.webrtcbin.emit('set-remote-description', desc, promise)
>>>>
>>>> 2c) It creates an answer calling `self.create_answer()`
>>>>
>>>> def create_answer(self):
>>>> promise = Gst.Promise.new_with_change_func(self.on_answer_created, self.webrtcbin, None)
>>>> self.webrtcbin.emit('create-answer', None, promise)
>>>>
>>>> def on_answer_created(self, promise, _, __):
>>>> ''' Local WebRTC stack has created an answer '''
>>>> promise.wait()
>>>> reply = promise.get_reply()
>>>> answer = reply.get_value('answer')
>>>> promise = Gst.Promise.new()
>>>> self.webrtcbin.emit('set-local-description', answer, promise)
>>>> promise.interrupt()
>>>> self.send_sdp_answer(answer) # Forwarding the answer to remote using the signaling server
>>>>
>>>> Observations so far:
>>>>
>>>> - It doesn't work if I don't set a payload-type which matches a payload type carried by the OFFERER or if I omit the payload-type. With ALL browsers it is sufficient to use the payload-type carried by the first H.264 element in the OFFER
>>>> - It is not necessary to set the `profile-level-id`
>>>>
>>>> The best result I can achieve with this flow is a spinning wheel in the browser. No video.
>>>>
>>>> BUT:
>>>>
>>>> When I issue a call to `start_pipeline` somewhere shortly after the start (I use the moment, when I have a connection with the signaling server for this) with _exactly_the_same_payload_type_that_will_match_later_, I 100% have a video...
>>>>
>>>> This MUST be a side effect, because the log complains about the second `start_pipeline` (warnings, it seems to ignore it).... But it works. Of course this is not a working solution, because
>>>>
>>>> a) I would have to know the payload-type beforehand (it doesn't work if I use another payload-type in the first start)
>>>> b) I would have to parse the incoming SDP for the first H.264 (which is doable)
>>>>
>>>> What am I doing wrong?
>>>>
>>>>
>>>>
>>>>
>>>> def start_pipeline(self, payload_type):
>>>> videotestsrc = Gst.ElementFactory.make("videotestsrc", "videotestsrc")
>>>> videotestsrc.set_property("is-live", True)
>>>> videotestsrc.set_property("pattern", "smpte")
>>>>
>>>> videotestsrc_caps = Gst.caps_from_string("video/x-raw, width=1280, height=720")
>>>>
>>>> videotestsrc_caps_filter = Gst.ElementFactory.make("capsfilter")
>>>> videotestsrc_caps_filter.set_property("caps", videotestsrc_caps)
>>>>
>>>> videoconvert = Gst.ElementFactory.make("videoconvert", "videoconvert")
>>>>
>>>> x264enc = Gst.ElementFactory.make("x264enc", "x264enc")
>>>> x264enc.set_property("threads", 4)
>>>>
>>>> x264enc_caps = Gst.caps_from_string("video/x-h264, profile=constrained-baseline")
>>>>
>>>> x264enc_caps_filter = Gst.ElementFactory.make("capsfilter")
>>>> x264enc_caps_filter.set_property("caps", x264enc_caps)
>>>>
>>>> rtph264pay = Gst.ElementFactory.make("rtph264pay", "rtph264pay")
>>>> rtph264pay.set_property("config-interval", -1)
>>>> rtph264pay.set_property("aggregate-mode", "zero-latency")
>>>>
>>>> rtph264pay_caps = Gst.caps_from_string("application/x-rtp, media=video, encoding-name=H264")
>>>> rtph264pay_caps.set_value("payload", payload_type)
>>>>
>>>> rtph264pay_caps_filter = Gst.ElementFactory.make("capsfilter")
>>>> rtph264pay_caps_filter.set_property("caps", rtph264pay_caps)
>>>>
>>>> self.pipeline.add(videotestsrc)
>>>> self.pipeline.add(videotestsrc_caps_filter)
>>>> self.pipeline.add(videoconvert)
>>>> self.pipeline.add(x264enc)
>>>> self.pipeline.add(x264enc_caps_filter)
>>>> self.pipeline.add(rtph264pay)
>>>> self.pipeline.add(rtph264pay_caps_filter)
>>>>
>>>> Gst.Element.link <http://gst.element.link/>(videotestsrc, videotestsrc_caps_filter)
>>>> Gst.Element.link <http://gst.element.link/>(videotestsrc_caps_filter, videoconvert)
>>>> Gst.Element.link <http://gst.element.link/>(videoconvert, x264enc)
>>>> Gst.Element.link <http://gst.element.link/>(x264enc, x264enc_caps_filter)
>>>> Gst.Element.link <http://gst.element.link/>(x264enc_caps_filter, rtph264pay)
>>>> Gst.Element.link <http://gst.element.link/>(rtph264pay, rtph264pay_caps_filter)
>>>> Gst.Element.link <http://gst.element.link/>(rtph264pay_caps_filter, self.webrtcbin)
>>>> self.pipeline.set_state(Gst.State.PLAYING)
>>>>
>>>
>>
>
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