webrtc: Padname src_0 is not unique in element webrtcbin0, not adding

[Anton Pryima]

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> 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/
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