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

Joshua C. Colp jcolp at digium.com
Wed Jun 12 08:24:55 CDT 2019


On Wed, Jun 12, 2019, at 10:17 AM, Abhay Gupta wrote:
> 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 . 

A quick glance shows the problem may be fragmentation related, at least for the original user. I'm actively working on such a thing and it is being tracked on another issue[1]. For your specific case it may or may not be the same.

[1] https://issues.asterisk.org/jira/browse/ASTERISK-28018

-- 
Joshua C. Colp
Digium - A Sangoma Company | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org



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