Re: [asterisk-users] Ast 13.10 to 13.11 stop working webrtc

the issue is fixed in current trunk head version El jue., 6 de oct. de 2016 a la(s) 12:07, Sebastian escribió: > the issue is with chan_sip not on rtp I will check wich commit break this > and fill an issue. > > > El mié., 5 de oct. de 2016 a la(s) 17:41, Sebastian > escribió: > > From this cha

Re: [asterisk-users] Ast 13.10 to 13.11 stop working webrtc

the issue is with chan_sip not on rtp I will check wich commit break this and fill an issue. El mié., 5 de oct. de 2016 a la(s) 17:41, Sebastian escribió: > From this change (res_rtp_asterisk): ast 13.10 to 13.11 webrtc JSSIP stop > working, failing with > > chan_sip.c:4083 retrans_pkt: Hanging

[asterisk-users] Ast 13.10 to 13.11 stop working webrtc

>From this change (res_rtp_asterisk): ast 13.10 to 13.11 webrtc JSSIP stop working, failing with chan_sip.c:4083 retrans_pkt: Hanging up call 7238b48c11581d4166b899bf747a05f7@130.211.62.184:0 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).