Helllo,
Greetings...
Being a novice person in WebRTC development area, I am trying to use Gstreamer based DTLS SRTP library for DTLS/SRTP support in WebRTC server. Client is an android based application on mobile.
I have followed the README file specified in source code directory in DTLS/SRTP implementation, ie.
1) decode pipeline starts first compared encode pipleline
2) unique connection-id parameter between DTLS encode/decode pipeline.
3) is-client property is set as FALSE since it is DTLS server.
4) PEM file being generated internally by Gstreamer DTLS library.
With the above settings, I do observed SSL_do_handshake returns zero(SYSCALL_ERROR), instead of one.
Did anyone have seen this error in WebRTC server implementation ?
Do I need any more settings in Gstreamer based DTLS/SRTP application for WebRTC server product?
Please do share any valid input to overcome this error.
Thank you.
Regards
Venkatesh
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