changing rtp capabilities

[Chuck Crisler]

The generated capabilities aren't random, they are default, except for the
seqnum-base, which RTP says can (should?) be random. Different elements
require different capabilities, so these may not hold for your case.
However, you can always initialize the capabilities like this:

            GstCaps *pSinkCaps = gst_caps_new_simple("application/x-rtp",
"media", G_TYPE_STRING, "video",
                     "encoding-name", G_TYPE_STRING, "H264", "payload",
G_TYPE_INT, 96,
                     "clock-rate", G_TYPE_INT, 90000, NULL);

            // Link the pipeline together.
            if (gst_element_link_filtered(source_, queue_, pSinkCaps)) {
            ...
            }
            gst_caps_unref(pSinkCaps);


On Wed, Feb 20, 2013 at 10:05 AM, Saurabh Sabnis <sausab at gmail.com> wrote:

> Hi,
>
> I am new to *gstreamer* and I am trying to send RTP stream over the
> network. The *'rtpvp8pay'* element automatically generates some random
> capabilities such as *seqnum-base, clock-rate* and *ssrc* that need to be
> used as cap filter for *'udpsrc'* element at the receiving side.
> Is it possible to manually set these capabilities at the sender side
> instead of letting 'rtpvp8pay' element to generate these randomly?
>
>
> Thanks and regards,
> --
> *Saurabh Sabnis*
> *Graduate Student*
> *College of Computing*
> *Georgia Institute of Technology*
> *Atlanta, GA 30332-0280*
> *ssabnis3 at gatech.edu*
>
> _______________________________________________
> gstreamer-devel mailing list
> gstreamer-devel at lists.freedesktop.org
> http://lists.freedesktop.org/mailman/listinfo/gstreamer-devel
>
>
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