Unencrypted SRTP fails with gstreamer gstsrtp

[dwmw2]

Since commit f1d1e57 using gstrtsp (in Pidgin, with the Pidgin-chime audio stream) fails.

(02:21:06) backend-fs2: farstream-send-codec-changed: codec: 97: audio CHIME clock:16000 channels:1
(02:21:06) mediamanager: gst pipeline error: Could not initialize SRTP encoder
(02:21:06) mediamanager: Debug details: ../ext/srtp/gstsrtpenc.c(1090): gst_srtp_enc_check_set_caps (): /GstPipeline:pipeline0/GstBin:conf_0x5c662637cc50/FsRtpConference:fsrtpconference0/GstRtpBin:rtpbin0/GstSrtpEnc:srtpenc_1:
Failed to add stream to SRTP encoder (err: 2)
(02:21:06) backend-fs2: gst error Could not initialize SRTP encoder

https://github.com/GStreamer/gst-plugins-bad/blob/master/ext/srtp/gstsrtp.c

[pabuhler]

@dwmw2, thanks for reporting, I will do some investigation and get back to you.

[pabuhler]

I have tried to look at both bits of code, and I see there is a miss match. The origin code in libSRTP was in correct as the cipher_key_len is also the size of the mast key and salt which is used to derive authentications keys, so the previous value of 16 is just incorrect. Exposing the NULL cipher and srtp_sec_serv_t in the public API is a bit of a double up and slightly confusing.
I am not sure what todo here, it is possible to revert the change and go back refactor why it was done in the first place. Alternatively we can make a patch on GStreamer / gstsrtp.c to set the right key size. @bifurcation do you remember the reason for this change?