Sebastian Damm wrote: > Asterisk gets a call from PSTN (via Sangoma PRI card). It sends out an > INVITE to the appropriate SIP peer. The SIP peer answers with 180 > without SDP, but still sends some RTP packets (silent). I don't know why > it does so, but it does. Asterisk now forwards the RTP packets to PSTN, > thus does not generate a Ringing itself. So for the caller the line is > silent until the callee picks up. > > According to the developers of the SIP PBX, Asterisk should ignore the > RTP packets, because there is no early media session established. To me, > this sounds correct. > > Is this a bug in Asterisk? Or is the behavior desired as it is? If not, > should I raise a bug in bugtracker?
No, this is incorrect. Asterisk sent an INVITE to the phone that included SDP, which means that Asterisk is willing to receive media from the phone and the phone is free to send it. The phone's response without SDP means Asterisk should not send audio *to* the phone, but it does not impact Asterisk receiving audio. -- Kevin P. Fleming Director of Software Technologies Digium, Inc. - "The Genuine Asterisk Experience" (TM) _______________________________________________ --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
