[gst-devel] RTP+MJPEG - Sniffing packets with wireshark, How clients join the rtp server

Marco Ballesio gibrovacco at gmail.com
Wed Dec 22 09:14:01 CET 2010


Hi,

one note I forgot to write before (sorry for the spam)..

On Wed, Dec 22, 2010 at 10:03 AM, Marco Ballesio <gibrovacco at gmail.com> wrote:

..snip..

>> 2) I do not understand very well how the client (player) join the RTP
>> server (streamer). I read the general RFC 1889. In the section about
>> RTP Control Protocol, it is not clear how the initialization works.
>
> RTCP is not about session initiation, as RFC1889 says, it is "to allow
> monitoring of the data delivery".  Read above for more details.

And also, unless you're dealing with very old HW/SW, you should use
RFC 3550 in place of RFC 1889 (not that changes are so drastic
though).

Regards

>
>> I
>> mean, when I start the streamer, does it start to send RTP+MJPEG
>> packets to the unicast/multicast destination even if nobody is
>> listening to it?
>
> Yes, it does. How could it happen without signaling otherwise?
>
>> Or does it start sending packets after the reception
>> of a particular packet from the client ? I read about feedbacks
>> packets from sender/receiver and about the BYE (Goodbye RTCP packet).
>> However, I have not found a sort of JOIN packet.
>>
>
> Because -unless you're using means other than RFC1889- there aren't
> any. Probably VLC performs a session initiation (I guess RTSP, you
> should see it in the network capture). If you need such a feature, on
> the GStreamer side you'll need to use the "rtspsrc" element.
>
> Regards
>
>>
>> Thanks in advance for the replies,
>>
>> Mauro
>>
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>




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