On 24.01.19 at 14:31 Joshua C. Colp wrote: > On Thu, Jan 24, 2019, at 9:23 AM, Michael Maier wrote: >> On 24.01.19 at 12:08 Joshua C. Colp wrote: >>> On Thu, Jan 24, 2019, at 7:00 AM, Michael Maier wrote: >>>> Hello! >>>> >>>> Given is an outbound call to a callee via asterisk to the ISP. After >>>> INVITE, the ISP sends 100 Trying and some time later 180 Ringing *w/* >>>> SDP and the header >>>> "P-Early-Media: sendonly" (no 183 or any other thing until callee >>>> responds). SDP in 180 contains media attribute *sendrecv* >>>> (contradicting to the P-Early-Media header). >>>> >>>> Asterisk 13.24.1 sends PRACK to the ISP (w/o SDP) and the 180 to the >>>> caller - but SDP is dropped and therefor no early media is possible! >>>> >>>> Is this behavior a bug or a feature? >>> >>> It would be a bug. That's not a scenario that I think anyone has really >>> tested or fully scoped out. Is media actually received while in the 180 >>> with Ringing stage? >> >> Yes - there is actually media been sent by the ISP, which is completely >> ignored by asterisk: >> >> [2019-01-24 13:24:20] DEBUG[3341][C-00000026]: res_rtp_asterisk.c:4109 >> ast_rtp_write: No remote address on RTP instance '0x7fedc40586d8' so >> dropping frame >> >> At this point, there isn't any media flowing between caller and >> asterisk and from asterisk to callee - just the ISP sends media to >> asterisk. >> >> Media is sent to the caller by asterisk at the moment, the 200 OK w/ >> SDP package has been received after callee picked up the phone. >> Therefore, the first few syllables are >> lost and you have to ask the name of the callee again, e.g. > > I'd suggest filing an issue[1] with a trace and console output then. > > [1] https://issues.asterisk.org/jira >
https://issues.asterisk.org/jira/browse/ASTERISK-28261 -- _____________________________________________________________________ -- 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
