Hi Joshua, thanks for the quick reply. Change was needed as the other side don’t accept the call with the attr a=setup .... and i only can change my side ..., i was able to remove the attr now, thanks for the hint,
Roland > On 31 May 2019, at 17:37, Joshua C. Colp <jc...@digium.com> wrote: > >> On Thu, May 30, 2019, at 9:25 PM, Learn&Use wrote: >> Hi all, >> >> I would need to remove the sdp attribute a=setup from INVITE’s going >> out to a SIP trunk ISP. I’m using chan_PJSIP with DTLS media encryption >> and was searching through various source files(res_pjsip_sdp_rtp,...) >> to make a change in the code to prevent that this attribute gets send >> out. I could not spot the correct location so far? I found it in >> chan_SIP but not chan_PJSIP. Would appreciate any hints, > > It is done in the SDP code in res_pjsip_sdp_rtp.c. Why do you need it removed > in the first place? > > -- > 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
