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I am trying to implement a Full HD video conferencing solution on a Raspberry Pi 3. As I only need the the latency I just wanted to create one simple pipeline for audio and video. I compiled gstreamer 1.13.0.1 from source, as the version 1.4.4 in the Raspian repository is to old (had problems with G.722 encoding). My pipeline for encryption seems to work - but my pipeline for decryption fails:

Encryption (runs):

gst-launch-1.0 v4l2src ! \ 'video/x-h264,width=1920,height=1080,framerate=30/1' ! \ h264parse config-interval=1 ! \ rtph264pay ! 'application/x-rtp, payload=(int)96, ssrc=(uint)1356955624' ! srtpenc key="012345678901234567890123456789012345678901234567890123456789" ! udpsink host=10.0.0.132 port=5000

Decryption: Decode:

gst-launch-1.0 udpsrc port=5000 ! 'application/x-rtp,encoding-name=(string)H264,ssrc=(uint)1356955624,srtp-key=(buffer)012345678901234567890123456789012345678901234567890123456789,srtp-cipher=(string)aes-128-icm,srtp-auth=(string)hmac-sha1-80' ! srtpdec ! rtph264depay ! h264parse ! avdec_h264 ! autovideosink



0:00:00.234264235 26462 0x156b380 ERROR GST_PIPELINE grammar.y:726:gst_parse_perform_link: could not link udpsrc0 to srtpdec0, srtpdec0 can't handle caps application/x-rtp, encoding-name=(string)H264, ssrc=(uint)1356955624, srtp-key=(buffer)012345678901234567890123456789012345678901234567890123456789, srtp-cipher=(string)aes-128-icm, srtp-auth=(string)hmac-sha1-80

So I think I need some additional caps - but which?

  • Just changed application/x-rtp to application/x-srtp .... sometimes it can be that easy.. ;-) – maunza Jun 17 '17 at 18:20
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I just changed application/x-rtp to application/x-srtp and it worked.

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