<html><head></head><body><div style="color:#000; background-color:#fff; font-family:Helvetica Neue-Light, Helvetica Neue Light, Helvetica Neue, Helvetica, Arial, Lucida Grande, Sans-Serif;font-size:16px"><div id="yui_3_16_0_1_1457372534592_2674">Hello,</div><div id="yui_3_16_0_1_1457372534592_2703"> </div><div id="yui_3_16_0_1_1457372534592_2675">In my current project, I am tasked to develop gstreamer based WebRTC server with SRTP support. As a novice person in this area, I am looking for some guidelines or hints to develop the product. But I do not find any hints in internet for a sample gstreamer WebRTC server with SRTP support. </div><div id="yui_3_16_0_1_1457372534592_2677">I have used gstreamer bad plugin for SRTP decode and encode functions along with Cisco's libsrtp. I am not able to setup gstreamer pipeline as auth tag comparison fails after extracting from SRTP stream. LibSRTP fails in comparing RTP auth tags. </div><div id="yui_3_16_0_1_1457372534592_2678">In RTPbin documentation, there was a mention of four additional gstreamer signals, but bad plugin do not have mention of it. </div><div id="yui_3_16_0_1_1457372534592_2679">The text snippet from the Gstreamer documentation is as below.</div><div id="yui_3_16_0_1_1457372534592_2680">GstRtpBin also has signals (“request-rtp-encoder”, “request-rtp-decoder”, “request-rtcp-encoder” and “request-rtp-decoder”) to dynamically request for RTP and RTCP encoders and decoders in order to support SRTP. The encoders must provide the pads rtp_sink_%u and rtp_src_%u for RTP and rtcp_sink_%u and rtcp_src_%u for RTCP. The session number will be used in the pad name. The decoders must provide rtp_sink and rtp_src for RTP and rtcp_sink and rtcp_src for RTCP. The decoders will be placed before the GstRtpSession element, thus they must support SSRC demuxing internally.</div><div id="yui_3_16_0_1_1457372534592_2681"><br id="yui_3_16_0_1_1457372534592_2682">My queries are as below:<br id="yui_3_16_0_1_1457372534592_2683">1. Do we have any good plugin for SRTP decode or encode function in Gstreamer? Please let me know.<br id="yui_3_16_0_1_1457372534592_2684">2. What are the dependent library for libSRTP and bad plugin to work for gstreamer SRTP based pipeline?<br id="yui_3_16_0_1_1457372534592_2685">3. How the graph looks like in successful scenario of Gstreamer based SRTP-only pipeline?</div><div id="yui_3_16_0_1_1457372534592_2686"><br id="yui_3_16_0_1_1457372534592_2687">Please share if any helpful info that you know or come across. Thank you in advance</div><div id="yui_3_16_0_1_1457372534592_2688" dir="ltr">Warm Regards<br id="yui_3_16_0_1_1457372534592_2689">Venkatesh<br></div></div></body></html>