webrtc: Padname src_0 is not unique in element webrtcbin0, not adding
Matthew Waters
ystreet00 at gmail.com
Sun Jan 10 13:21:35 UTC 2021
webrtcbin currently supports some reconfiguration scenarios. Format
changing (e.g. H264 to VP8) however is not one of those scenarios.
Cheers
-Matt
On 10/1/21 9:31 pm, Anton Pryima wrote:
> Hello,
> As for now, as I understand, gstwebrtcbin is not support live
> reconfiguration. So, I was managed to change streams from one to
> another but it must have the same format (if initial negotiation was
> for ex. h264 you can change stream to any h264).
> To any other format, you need to reestablish connection again.
>
> Best regards,
> Anton.
>
> On Sat, Jan 9, 2021, 15:00 coreykernel <corey at kernellabs.io
> <mailto:corey at kernellabs.io>> wrote:
>
> Hey Anton,
>
> Did you ever resolve your issue. I'm getting a similar mline error
> when I
> try to send a 2nd offer after adding the new stream:
>
> webrtc_sink = gst_element_get_request_pad(webrtc, "sink_1");
> ret = gst_pad_link(q3_src, webrtc_sink);
> g_assert_cmphex(ret, ==, GST_PAD_LINK_OK);
> promise = gst_promise_new_with_change_func(on_offer_created, NULL,
> NULL);
> g_signal_emit_by_name(G_OBJECT(webrtc), "create-offer", NULL,
> promise);
>
> The on_offer_created callback is never called and I get an error:
>
> ERROR:../ext/webrtc/gstwebrtcbin.c:2304:sdp_media_from_transceiver:
> assertion failed: (trans->mline == -1 || trans->mline == media_idx)
> Bail out!
> ERROR:../ext/webrtc/gstwebrtcbin.c:2304:sdp_media_from_transceiver:
> assertion failed: (trans->mline == -1 || trans->mline == media_idx)
>
> Do I need to remove the original stream that I am replacing? Or is
> it OK to
> send two streams to the peer?
>
>
>
> --
> Sent from: http://gstreamer-devel.966125.n4.nabble.com/
> <http://gstreamer-devel.966125.n4.nabble.com/>
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