Hello all, I took a look at this patch and it works fine when I tested with STUN. I made some tests with a simple NAT and I hadn't any problems. I think it's a very important feature to be added on the next 0.10.x release. Regards, Birunko <div><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">From: Kai Vehmanen <<a href="mailto:kvehmanen@eca.cx" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)"> kvehmanen@eca.cx</a>> Date: Nov 30, 2005 3:33 PM Subject: [gst-devel] PATCH: sockfd property to udpsrc/dynudpsink elements To: <a href="mailto:gstreamer-devel@lists.sourceforge.net" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)"> gstreamer-devel@lists.sourceforge.net </a> Hello all, here's a patch that adds a 'sockfd' property to the udpsrc/dynudpsink elements in gst-plugins-good. I'd like to get these into 0.10 if possible (sorry for the bad/late timing). If 0.10 is out of the question, when would be an appropriate time? I assume adding new properties to elements is possible in 0.10.x releases). Comments on the patch are of course also very much welcome. Anyways, this patch is needed for two+ reasons: * To be able to send&receive RTP+RTCP when connected to network    via a NAT/FW device, the same local port must be used for sending and    receiving (-> symmetric RTP). This is an essential feature for any VoIP    style client using RTP. * The current udpsrc/dynudpsink interface is clumsy if the    requested port is reserved at NULL->READY state change. And especially    if you need to reserve paired ports on a busy system (X for RTP, and    X+1 for RTCP) without race-conditions. It would be much easier to have    the app reserve and bind to the ports, and then pass the bound sockfds to    udpsrc/dynudpsink for use. * Longer term: to integrate NAT traversal mechanisms such as STUN    <<a href="http://www.faqs.org/rfcs/rfc3489.html" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">http://www.faqs.org/rfcs/rfc3489.html</a>> and UPnP-IGD to gstreamer, the    STUN client API has to create and bind the socket (you cannot do    this inside udpsrc/dynudpsink). For testing, I've committed patches to the Farsight darcs repository, and modified the gsmsendrecv.c test program to utilize the patched dynudpsink and udpsrc (see the below message). This allows to symmetric use of both RTP and RTCP (verified to work with current gstreamer CVS). You can get the needed elements with:    darcs get <a href="http://projects.collabora.co.uk/darcs/farsight/gst-plugins-farsight" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)"> http://projects.collabora.co.uk/darcs/farsight/gst-plugins-farsight </a> ---------- Forwarded message ---------- Date: Wed, 30 Nov 2005 20:12:04 +0200 (EET) From: Kai Vehmanen <<a href="mailto:kvehmanen@eca.cx" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)"> kvehmanen@eca.cx</a>> To: <a href="mailto:farsight-devel@lists.sourceforge.net" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)"> farsight-devel@lists.sourceforge.net</a> Subject: [Farsight-devel] darcs-commits; symmetric-rtp, accepting own packets Hello, I just committed some changes to enable symmetric RTP support: * Added 'rtp_sockfd' and 'rtcp_sockfd' properties to rtpbin.    - If not specified, rtpbin will behave as before.    - Requires the patches to udpsrc and dynudpsink (will      send to gstreamer-devel). * Configure jrtplib to accept own packets by default.    - This allows to use the test apps in gst-plugins-farsight      without patching jrtplib, and without getenv() hacks.    - If someone really wants to use multicast with rtpbin,      please go ahead and add a nice way to select this      at rtpbin creation time. * Modified gst-plugins-farsight/tests/gsmsendrecv.c    - Creates and binds sockets for RTP and RTCP, passes      these to rtpbin, and runs a RTP/GSM test over      local loopback.    - Tested to work with a snapshot of today's (2005-11-30)      gstreamer/gst-plugins-* CVS.    - Thanks to the accept-own-packets change above,      you can actually run the test without hacking'n'patching      jrtplib as you had to do before. --   under work: Sofia-SIP at <a href="http://sofia-sip.sf.net" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">http://sofia-sip.sf.net</a> </blockquote></div> -- current->id = 0; >> Powered by Linux