[asterisk-dev] Issue of DTLS setup issue in some webrtc calls

[Abhay Gupta]

I was looking at the bug ASTERISK-27826 and found that in file res_rtp_asterisk.c  and function __rtp_recvfrom in call with the issue ssl state is  "before/accept initialization"  and in successful cases the state is  "SSL negotiation finished successfully" 

The function is called twice in block "if ((*in >= 20) && (*in <= 63))" wherein first instance the state is unknown and then if negotiation is successful the call is fine and if it remains in before/accept initialisation the same is never called again and results in no voice .

How can we ensure that SSL negotiation is successful in all cases . 

   
Regards,

Abhay 








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