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

[Abhay Gupta]

Ok seems that in that particular case of ASTERISK-28018 all calls to a ISP fails but in my case only 1-2% calls get this issue and in that case garbage is recorded in asterisk mix monitor and chrome shows Failed to unprotect RTP packet and this is similar to ASTERISK-27826

I also understand that before/accept initialization is a intermediate signalling state so I will try on test setup with a for loop of 5 times with 1 microsecond sleep and see if the state changes .

Regards

> On 12-Jun-2019, at 6:54 PM, Joshua C. Colp <jcolp at digium.com> wrote:
> 
> 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
> 
> -- 
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> 
> asterisk-dev mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-dev