<P>Hello All,</P>
<P> So lets summerise our work on GstRtp plugins, which we currently base on gphone's rtp implementation named librtp. The rtpsend was first written by zaheer, but there was no rtprecv. So i & wtay started working on it & soon realized that rtpsend also needs a lot of change. So we made progress & now we have the rtp plugins streaming. But the things didnt end up so simple. We wanted to support gsm, mpeg audio/video( layer 1,2 & 3 ),raw audio etc. All of these seem to work nice now but the mpeg audio layer 3( i havent got the chance to check layer 1 & 2 though ), which uses some forward refrences in its algorithem or something. </P>
<P>I added a packet-sequencing algorithem of mine too in the rtprecv that works but i think there is some bug in it too.</P>
<P>I was busy today making the mtu( maximum transfer unit ) in the rtpsend to change according to the payload type & tell the rtprecv about its current value along with all other things it already tell's rtprecv. I watched some 4-5 min. mpeg 1 videos with mtu=1024 today & it all worked fine. wtay: plz decide/change the mtu sizes accordingly in the newcaps() of rtpsend.</P><BR><BR><A href="http://zchat.sourceforge.net/Zeenix.html">ZEESHAN ALI</A><p><br><hr size=1><b>Do You Yahoo!?</b><br>
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