Ast 13.10 To 13.11 Stop Working Webrtc

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Asterisk Users 2 Comments

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).

is there any way to configure to have the previous behaviour?
Im trying to set dtlscipher

2 thoughts on - 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:

  • the issue is fixed in current trunk head version

    El jue., 6 de oct. de 2016 a la(s) 12:07, Sebastian
    escribió: