As I have checked the state remains in “before/accept initialisation” . For the time being I have hangup the call with such a issue , will wait for the fragmentation issue resolution and will test the same after such resolution .
Regards, > On 12-Jun-2019, at 7:35 PM, Abhay Gupta <ab...@avissol.com> wrote: > > 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 <jc...@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 > -- _____________________________________________________________________ -- 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
